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 ==============================================================================
10 ------------------------------------------------------------------------------
11 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
12 ------------------------------------------------------------------------------
19 * The application no longer does agent authentication. The dialplan needs to
20 perform this function before running AgentLogin. If the agent is already
21 logged in, dialplan will continue with the AGENT_STATUS channel variable
22 set to ALREADY_LOGGED_IN.
26 * Application removed. It was a holdover from when AgentCallbackLogin was
31 * All participants in a bridge can now be kicked out of a conference room
32 by specifying the channel parameter as 'all' in the ConfBridge kick CLI
33 command, i.e., "confbridge kick <conference> all"
37 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
40 * Variables are no longer purged from the original CDR. See the 'v' option for
43 * The 'A' option has been removed. The Answer time on a CDR is never updated
46 * The 'd' option has been removed. The disposition on a CDR is a function of
47 the state of the channel and cannot be altered.
49 * The 'D' option has been removed. Who the Party B is on a CDR is a function
50 of the state of the respective channels, and cannot be altered.
52 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
53 such that the start time and, if applicable, the answer time was updated.
54 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
55 'r' option now triggers the Reset, setting the start time (and answer time
56 if applicable) to the current time.
58 * The 's' option has been removed. A variable can be set on the original CDR
59 if desired using the CDR function, and removed from a forked CDR using the
62 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
63 longer applies in the CDR engine.
65 * The 'v' option now prevents the copy of the variables from the original CDR
66 to the forked CDR. Previously the variables were always copied but were
67 removed from the original. Removing variables from a CDR can have unintended
68 side effects - this option allows the user to prevent propagation of
69 variables from the original to the forked without modifying the original.
73 * Added the 'n' option to MeetMe to prevent application of the DENOISE function
74 to a channel joining a conference. Some channel drivers that vary the number
75 of audio samples in a voice frame will experience significant quality problems
76 if a denoiser is attached to the channel; this option gives them the ability
77 to remove the denoiser without having to unload func_speex.
81 * The NoCDR application is deprecated. Please use the CDR_PROP function to
83 * While the NoCDR application will prevent CDRs for a channel from being
84 propagated to registered CDR backends, it will not prevent that data from
85 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
86 function that enables CDRs on a channel will restore those records that have
87 not yet been finalized.
91 * Add queue available hint. exten => 8501,hint,Queue:markq_avail
92 Note: the suffix '_avail' after the queuename.
93 Reports 'InUse' for no logged in agents or no free agents.
94 Reports 'Idle' when an agent is free.
96 * The configuration options eventwhencalled and eventmemberstatus have been
97 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
98 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
99 sent. The "Variable" fields will also no longer exist on the Agent* events.
101 * The queue log now differentiates between blind and attended transfers. A
102 blind transfer will result in a BLINDTRANSFER message with the destination
103 context and extension. An attended transfer will result in an
104 ATTENDEDTRANSFER message. This message will indicate the method by which
105 the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
106 for running an application on a bridge or channel, or "LINK" for linking
107 two bridges together with local channels.
109 * Queues now support a hint for member paused state. The hint uses the form
110 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
111 are the name of the queue and the name of the member to subscribe to,
112 respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
113 Members will show as In Use when paused.
117 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
118 CDRs when they were previously disabled on a channel.
119 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
120 backends occurs on an as-needed basis in order to preserve linkedid
121 propagation and other needed behavior.
125 * This application is deprecated in favor of the CHANNEL function.
129 * UserEvent will now handle duplicate keys by overwriting the previous value
130 assigned to the key. UserEvent invocations will also be distributed to any
131 interested res_stasis applications.
136 * Asterisk now optionally uses libxslt to improve XML documentation generation
137 and maintainability. If libxslt is not available on the system, some XML
138 documentation will be incomplete.
143 * Redirecting reasons can now be set to arbitrary strings. This means
144 that the REDIRECTING dialplan function can be used to set the redirecting
145 reason to any string. It also allows for custom strings to be read as the
146 redirecting reason from SIP Diversion headers.
148 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
149 must be on the channel initiating the transfer to have any effect.
151 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
152 driver specific. If the channel variable is set on the transferrer channel,
153 the sound will be played to the target of an attended transfer.
155 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
156 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
157 listed. Any more peers in the bridge will not be included in the list.
158 BRIDGEPEER is not valid in holding bridges like parking since those channels
159 do not talk to each other even though they are in a bridge.
161 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
162 and will contain a value if the BRIDGEPEER's channel driver supports it.
164 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
165 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
166 activated the dynamic feature.
168 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
169 only on the channel executing the dynamic feature. Executing a dynamic
170 feature on the bridge peer in a multi-party bridge will execute it on all
171 peers of the activating channel.
173 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
174 was responsible for an attended transfer in a similar fashion to
180 * Added pass through support for VP8 and Opus
182 * Added format attribute negotiation for the Opus codec. Format attribute
183 negotiation is provided by the res_format_attr_opus module.
186 AMI (Asterisk Manager Interface)
188 * The SIPshowpeer action will now include a 'SubscribeContext' field for a peer
189 in its response if the peer has a subscribe context set.
191 * The SIPqualifypeer action now acknowledges the request once it has established
192 that the request is against a known peer. It also issues a new event,
193 'SIPQualifyPeerDone', once the qualify action has been completed.
195 * The PlayDTMF action now supports an optional 'Duration' parameter. This
196 specifies the duration of the digit to be played, in milliseconds.
198 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
199 updates when changes occur instead of requiring the use of pollmailboxes.
201 * CLI Command 'Manager Show Commands' no longer truncates command names longer
202 than 15 characters and no longer shows authorization requirement for commands.
203 'Manager Show Command' now displays the privileges needed for using a given
204 manager command instead.
206 * Added new action "ControlPlayback". The ControlPlayback action allows an AMI
207 client to manipulate audio currently being played back on a channel. The
208 supported operations depend on the application being used to send audio to
209 the channel. When the audio playback was initiated using the ControlPlayback
210 application or CONTROL STREAM FILE AGI command, the audio can be paused,
211 stopped, restarted, reversed, or skipped forward. When initiated by other
212 mechanisms (such as the Playback application), the audio can be stopped,
213 reversed, or skipped forward.
215 * Channel related events now contain a snapshot of channel state, adding new
216 fields to many of these events.
218 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
219 in a future release. Please use the common 'Exten' field instead.
221 * The AMI event 'UserEvent' from app_userevent now contains the channel state
222 fields. The channel state fields will come before the body fields.
224 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
225 'UnParkedCall' have changed significantly in the new res_parking module.
227 The 'Channel' and 'From' headers are gone. For the channel that was parked
228 or is coming out of parking, a 'Parkee' channel snapshot is issued and it
229 has a number of fields associated with it. The old 'Channel' header relayed
230 the same data as the new 'ParkeeChannel' header.
232 The 'From' field was ambiguous and changed meaning depending on the event.
233 for most of these, it was the name of the channel that parked the call
234 (the 'Parker'). There is no longer a header that provides this channel name,
235 however the 'ParkerDialString' will contain a dialstring to redial the
236 device that parked the call.
238 On UnParkedCall events, the 'From' header would instead represent the
239 channel responsible for retrieving the parkee. It receives a channel
240 snapshot labeled 'Retriever'. The 'from' field is is replaced with
243 Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
245 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
246 fashion has changed the field names 'StartExten' and 'StopExten' to
247 'StartSpace' and 'StopSpace' respectively.
249 * The deprecated use of | (pipe) as a separator in the channelvars setting in
250 manager.conf has been removed.
252 * Channel Variables conveyed with a channel no longer contain the name of the
253 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
254 ChanVariable: bar=baz. When multiple channels are present in a single AMI
255 event, the various ChanVariable fields will contain a suffix that specifies
256 which channel they correspond to.
258 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
259 event always conveys the AMI event for a particular channel.
261 * All "Reload" events have been consolidated into a single event type. This
262 event will always contain a Module field specifying the name of the module
263 and a Status field denoting the result of the reload. All modules now issue
264 this event when being reloaded.
266 * The "ModuleLoadReport" event has been removed. Most AMI connections would
267 fail to receive this event due to being connected after modules have loaded.
268 AMI connections that want to know when Asterisk is ready should listen for
269 the "FullyBooted" event.
271 * app_fax now sends the same send fax/receive fax events as res_fax. The
272 "FaxSent" event is now the "SendFAX" event, and the "FaxReceived" event is
273 now the "ReceiveFAX" event.
275 * The MusicOnHold event is now two events: MusicOnHoldStart and
276 MusicOnHoldStop. The sub type field has been removed.
278 * The JabberEvent event has been removed. It is not AMI's purpose to be a
279 carrier for another protocol.
281 * The Bridge Manager action's Playtone header now accepts more fine-grained
282 options. "Channel1" and "Channel2" may be specified in order to play a tone
283 to the specific channel. "Both" may be specified to play a tone to both
284 channels. The old "yes" option is still accepted as a way of playing the
285 tone to Channel2 only.
287 * The AMI 'Status' response event to the AMI Status action replaces the
288 BridgedChannel and BridgedUniqueid headers with the BridgeID header to
289 indicate what bridge the channel is currently in.
291 * The AMI 'Hold' event has been moved out of individual channel drivers, into
292 core, and is now two events: Hold and Unhold. The status field has been
295 * The AMI events in app_queue have been made more consistent with each other.
296 Events that reference channels (QueueCaller* and Agent*) will show
297 information about each channel. The (infamous) "Join" and "Leave" AMI
298 events have been changed to "QueueCallerJoin" and "QueueCallerLeave".
300 * The MCID AMI event now publishes a channel snapshot when available and
301 its non-channel-snapshot parameters now use either the "MCallerID" or
302 "MConnectedID" prefixes with Subaddr*, Name*, and Num* suffixes instead
303 of "CallerID" and "ConnectedID" to avoid confusion with similarly named
304 parameters in the channel snapshot.
306 * The AMI events "Agentlogin" and "Agentlogoff" have been renamed
307 "AgentLogin" and "AgentLogoff" respectively.
309 * The "Channel" key used in the "AlarmClear", "Alarm", and "DNDState" has been
310 renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
312 * "ChannelUpdate" events have been removed.
314 * AMI events now contain a SystemName field, if available.
316 * Local channel optimization is now conveyed in two events:
317 LocalOptimizationBegin and LocalOptimizationEnd. The Begin event is sent
318 when the Local channel driver begins attempting to optimize itself out of
319 the media path; the End event is sent after the channel halves have
320 successfully optimized themselves out of the media path.
322 * Local channel information in events is now prefixed with "LocalOne" and
323 "LocalTwo". This replaces the suffix of "1" and "2" for the two halves of
324 the Local channel. This affects the LocalBridge, LocalOptimizationBegin,
325 and LocalOptimizationEnd events.
327 * The option 'allowmultiplelogin' can now be set or overriden in a particular
328 account. When set in the general context, it will act as the default
329 setting for defined accounts.
331 * The 'BridgeAction' event was removed. It technically added no value, as the
332 Bridge Action already receives confirmation of the bridge through a
333 successful completion Event.
335 * The 'BridgeExec' events were removed. These events duplicated the events that
336 occur in the Briding API, and are conveyed now through BridgeCreate,
337 BridgeEnter, and BridgeLeave events.
340 AGI (Asterisk Gateway Interface)
342 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
344 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
347 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
348 will start the playback of the audio at the position specified. It will
349 also return the final position of the file in 'endpos'.
351 * The SAY ALPHA command now accepts an additional parameter to control
352 whether it specifies the case of uppercase, lowercase, or all letters to
353 provide functionality similar to SayAlphaCase.
355 CDR (Call Detail Records)
357 * Significant changes have been made to the behavior of CDRs. For a full
358 definition of CDR behavior in Asterisk 12, please read the specification
359 on the Asterisk wiki (wiki.asterisk.org).
361 * CDRs will now be created between all participants in a bridge. For each
362 pair of channels in a bridge, a CDR is created to represent the path of
363 communication between those two endpoints. This lets an end user choose who
364 to bill for what during bridge operations with multiple parties.
366 * The duration, billsec, start, answer, and end times now reflect the times
367 associated with the current CDR for the channel, as opposed to a cumulative
368 measurement of all CDRs for that channel.
370 * When a CDR is dispatched, user defined CDR variables from both parties are
371 included in the resulting CDR. If both parties have the same variable, only
372 the Party A value is provided.
374 CEL (Channel Event Logging)
376 * The 'extra' field of all CEL events that use it now consists of a JSON blob
377 with key/value pairs which are defined in the Asterisk 12 CEL documentation.
379 * AST_CEL_BLINDTRANSFER events now report the transferee bridge unique
380 identifier, extension, and context in a JSON blob as the extra string
381 instead of the transferee channel name as the peer.
383 * AST_CEL_ATTENDEDTRANSFER events now report the peer as NULL and additional
384 information in the 'extra' string as a JSON blob. For transfers that occur
385 between two bridged channels, the 'extra' JSON blob contains the primary
386 bridge unique identifier, the secondary channel name, and the secondary
387 bridge unique identifier. For transfers that occur between a bridged channel
388 and a channel running an app, the 'extra' JSON blob contains the primary
389 bridge unique identifier, the secondary channel name, and the app name.
391 * AST_CEL_LOCAL_OPTIMIZE events have been added to convey local channel
392 optimizations with the record occurring for the semi-one channel and
393 the semi-two channel name in the peer field.
397 * The BRIDGE_FEATURES channel variable would previously only set features for
398 the calling party and would set this feature regardless of whether the
399 feature was in caps or in lowercase. Use of a caps feature for a letter
400 will now apply the feature to the calling party while use of a lowercase
401 letter will apply that feature to the called party.
403 * Add support for automixmon to the BRIDGE_FEATURES channel variable.
405 * Parking has been pulled from core and placed into a separate module called
406 res_parking. See Parking changes below for more details.
408 * You can now have the settings for a channel updated using the FEATURE()
409 and FEATUREMAP() functions inherited to child channels by setting
410 FEATURE(inherit)=yes.
412 * automixmon now supports additional channel variables from automon including:
413 TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
414 and TOUCH_MIXMONITOR_MESSAGE_STOP
416 * A new general features.conf option 'recordingfailsound' has been added which
417 allowssetting a failure sound for a user tries to invoke a recording feature
418 such as automon or automixmon and it fails.
422 * When performing queue pause/unpause on an interface without specifying an
423 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
424 least one member of any queue exists for that interface.
426 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
427 for realtime queue log entries.
431 * Parking is now implemented as a module instead of as core functionality.
432 The preferred way to configure parking is now through res_parking.conf while
433 configuration through features.conf is not currently supported.
435 * res_parking uses the configuration framework. If an invalid configuration is
436 supplied, res_parking will fail to load or fail to reload. Previously parking
437 lots that were misconfigured would generally be accepted with certain
438 configuration problems leading to individual disabled parking lots.
440 * Parked calls are now placed in bridges. This is a largely architectural change,
441 but it could have some implications in allowing for new parked call retrieval
442 methods and the contents of parking lots will be visible though certain bridge
445 * The order of arguments for the new parking applications are different from the
446 old ones to be more intuitive. Timeout and return context/exten/priority are now
447 implemented as options. parking_lot_name is now the first parameter. See the
448 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
449 in-depth information as well as syntax.
451 * Extensions are no longer automatically created in the dialplan to park calls,
452 pickup parked calls, etc by default.
454 * adsipark is no longer supported under the new parking model
456 * The PARKINGSLOT channel variable has been deprecated in favor of PARKING_SPACE
457 to match the naming scheme of the new system.
459 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
460 channel even when comebactoorigin=yes
462 * New CLI command 'parking show' allows you to inspect the currently in use
463 parking lots. 'parking show <parkinglot>' will also show the parked calls
464 in that specific parking lot.
466 * The CLI command 'parkedcalls' is now deprecated in favor of
467 'parking show <parkinglot>'.
469 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
470 can be used to get a list of parked calls only for a specific parking lot.
472 * The AMI command 'Park' has had the argument 'Channel2' renamed to
473 'TimeoutChannel'. 'TimeoutChannel' is no longer a required argument.
474 'Channel2' can still be used as the argument name, but it is deprecated
475 and the 'TimeoutChannel' argument will be used if both are present.
477 * The ParkAndAnnounce application is now provided through res_parking instead
478 of through the separate app_parkandannounce module.
480 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
481 by default. Instead, it will follow the timeout rules of the parking lot. The
482 old behavior can be reproduced by using the 'c' option.
484 * Dynamic parking lots will now fail to be created if the parking lot specified
485 by PARKINGDYNAMIC does not exist.
487 * Dynamic parking lots will also fail to be created now if they require exclusive
488 park and parkedcall extensions which overlap with other parking lots.
490 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
491 currently contain no calls. Dynamic parking lots containing parked calls will
492 persist through the reloads without alteration.
494 * If parkext_exclusive is set for a parking lot and that extension is already in
495 use when that parking lot tries to register it, this is now considered a parking
496 system configuration error. Configurations which do this will be rejected.
497 Dynamic parking lots which try to register extensions that already exist will
500 * Added a channel variable PARKER_FLAT which stores the name of the extension
501 that would be used to come back to if comebacktoorigin was set to use. This can
502 be useful when comebacktoorigin is off if you still want to use the extensions
503 in the park-dial context that are generated to redial the parker on timeout.
507 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
508 will store the path information for that peer when it registers. Realtime
509 tables can also use the 'supportpath' field to enable Path header support.
511 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
512 objectIdentifier. This maps to the supportpath option in sip.conf.
516 * All future modules which utilize Sorcery for object persistence must have a
517 column named "id" within their schema when using the Sorcery realtime module.
518 This column must be able to contain a string of up to 128 characters in length.
520 Security Events Framework
521 -------------------------
522 * Security Event timestamps now use ISO 8601 formatted date/time instead of the
523 "seconds-microseconds" format that it was using previously.
528 * When a channel driver is configured to enable jiterbuffers, they are now
529 applied unconditionally when a channel joins a bridge. If a jitterbuffer
530 is already set for that channel when it enters, such as by the JITTERBUFFER
531 function, then the existing jitterbuffer will be used and the one set by
532 the channel driver will not be applied.
536 * The updatecdr option has been removed. Altering the names of channels on a
537 CDR is not supported - the name of the channel is the name of the channel,
538 and pretending otherwise helps no one.
539 * The AGENTUPDATECDR channel variable has also been removed, for the same
540 reason as the updatecdr option.
541 * The driver is no longer a Data retrieval API data provider for the
543 * The endcall and enddtmf configuration options are removed. Use the
544 dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
545 channel before calling AgentLogin.
546 * chan_agent is removed and replaced with AgentLogin and AgentRequest dialplan
547 applications. Agents are connected with callers using the new AgentRequest
548 dialplan application. The Agents:<agent-id> device state is available to
549 monitor the status of an agent. See agents.conf.sample for valid
550 configuration options.
554 * chan_bridge is removed and its functionality is incorporated into ConfBridge
559 * The /b option is removed.
561 * chan_local moved into the system core and is no longer a loadable module.
565 * Added general support for busy detection.
567 * Added ECAM command support for Sony Ericsson phones.
571 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
572 using the 'supportpath' setting, either on a global basis or on a peer basis.
573 This setting enables Asterisk to route outgoing out-of-dialog requests via a
574 set of proxies by using a pre-loaded route-set defined by the Path headers in
575 the REGISTER request. See Realtime updates for more configuration information.
577 * The SIP_CODEC family of variables may now specify more than one codec. Each
578 codec must be separated by a comma. The first codec specified is the
579 preferred codec for the offer. This allows a dialplan writer to specify both
580 audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
587 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
588 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
589 The value of this setting is ignored when disabled is used for the argument.
593 * The 'amaflags' and 'accountcode' attributes for the CDR function are
594 deprecated. Use the CHANNEL function instead to access these attributes.
595 * The 'l' option has been removed. When reading a CDR attribute, the most
596 recent record is always used. When writing a CDR attribute, all non-finalized
598 * The 'r' option has been removed, for the same reason as the 'l' option.
599 * The 's' option has been removed, as LOCKED semantics no longer exist in the
604 * A new function CDR_PROP has been added. This function lets you set properties
605 on a channel's active CDRs. This function is write-only. Properties accept
606 boolean values to set/clear them on the channel's CDRs. Valid properties
608 * 'party_a' - make this channel the preferred Party A in any CDR between two
609 channels. If two channels have this property set, the creation time of the
610 channel is used to determine who is Party A. Note that dialed channels are
611 never Party A in a CDR.
612 * 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
613 application when set to True, and analogous to the 'e' option in ResetCDR
622 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
623 them, an Asterisk-specific version of pjproject needs to be installed.
624 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
628 * Device state for XMPP buddies is now available using the following format:
629 XMPP/<client name>/<buddy address>
630 If any resource is available the device state is considered to be not in use.
631 If no resources exist or all are unavailable the device state is considered
640 * The safe_asterisk script will now install over previously installations.
641 In previous versions of Asterisk, once installed a 'make install' would
642 skip over safe_asterisk if it was already installed.
643 * Certain options in safe_asterisk can now be configured from the
644 safe_asterisk.conf file. A sample version of this is located in the
647 ------------------------------------------------------------------------------
648 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
649 ------------------------------------------------------------------------------
655 * The Asterisk build system will now build and install a shared library
656 (libasteriskssl.so) used to wrap various initialization and shutdown functions
657 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
658 that Asterisk can ensure that these functions do *not* get called by any
659 modules that are loaded into Asterisk, since they should only be called once
660 in any single process. If desired, this feature can be disabled by supplying
661 the "--disable-asteriskssl" option to the configure script.
663 * A new make target, 'full', has been added to the Makefile. This performs
664 the same compilation actions as make all, but will also scan the entirety of
665 each source file for documentation. This option is needed to generate AMI
666 event documentation. Note that your system must have Python in order for
667 this make target to succeed.
669 * The optimization portion of the build system has been reworked to avoid
670 broken builds on certain architectures. All architecture-specific
671 optimization has been removed in favor of using -march=native to allow gcc
672 to detect the environment in which it is running when possible. This can
673 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
675 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
676 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
678 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
679 previously parsed the header file to obtain the version of Asterisk, you
680 will now have to go through Asterisk to get the version information.
688 * Added 'F()' option. Similar to the dial option, this can be supplied with
689 arguments indicating where the callee should go after the caller is hung up,
690 or without options specified, the priority after the Queue will be used.
695 * Added menu action admin_toggle_mute_participants. This will mute / unmute
696 all non-admin participants on a conference. The confbridge configuration
697 file also allows for the default sounds played to all conference users when
698 this occurs to be overriden using sound_participants_unmuted and
699 sound_participants_muted.
701 * Added menu action participant_count. This will playback the number of
702 current participants in a conference.
704 * Added announcement configuration option to user profile. If set the sound
705 file will be played to the user, and only the user, upon joining the
708 * Added record_file_append option that defaults to "yes", but if set to no
709 will create a new file between each start/stop recording.
714 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
715 channels respectively before the callee channels are called.
720 * Added support for IPv6.
722 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
723 external process will cause the current playlist to be cleared, including
724 stopping any audio file that is currently playing. This is useful when you
725 want to interrupt audio playback only when specific DTMF is entered by the
731 * A new option, 'I' has been added to app_followme. By setting this option,
732 Asterisk will not update the caller with connected line changes when they
733 occur. This is similar to app_dial and app_queue.
735 * The 'N' option is now ignored if the call is already answered.
737 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
738 and caller channels respectively before the callee channels are called.
740 * The winning FollowMe outgoing call is now put on hold if the caller put it on
746 * MixMonitor hooks now have IDs associated with them which can be used to
747 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
748 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
749 now accepts that ID as an argument.
751 * Added 'm' option, which stores a copy of the recording as a voicemail in the
757 * The connect action in app_mysql now allows you to specify a port number to
758 connect to. This is useful if you run a MySQL server on a non-standard
764 * Increased the default number of allowed destinations from 5 to 12.
769 * The app_page application now no longer depends on DAHDI or app_meetme. It
770 has been re-architected to use app_confbridge internally.
775 * Added queue options autopausebusy and autopauseunavail for automatically
776 pausing a queue member when their device reports busy or congestion.
778 * The 'ignorebusy' option for queue members has been deprecated in favor of
779 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
780 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
781 per interface basis. Individual ringinuse values can now be set in
782 queues.conf via an argument to member definitions. Lastly, the queue
783 'ringinuse' setting now only determines defaults for the per member
784 'ringinuse' setting and does not override per member settings like it does
787 * Added 'F()' option. Similar to the dial option, this can be supplied with
788 arguments indicating where the callee should go after the caller is hung up,
789 or without options specified, the priority after the Queue will be used.
791 * Added new option log_member_name_as_agent, which will cause the membername to
792 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
793 state_interface has been set.
795 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
797 * App_queue will now play periodic announcements for the caller that
798 holds the first position in the queue while waiting for answer.
802 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
803 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
804 changed arguments to SayUnixTime so that every option is truly optional even
805 when using multiple options (so that j option could be used without having to
806 manually specify timezone and format) There are other benefits, e.g., format
807 can now be used without specifying time zone as well.
812 * Addition of the VM_INFO function - see Function changes.
814 * The imapserver, imapport, and imapflags configuration options can now be
815 overriden on a user by user basis.
817 * When voicemail plays a message's envelope with saycid set to yes, when
818 reaching the caller id field it will play a recording of a file with the same
819 base name as the sender's callerid if there is a similarly named file in
820 <astspooldir>/recordings/callerids/
822 * Voicemails now contains a unique message identifier "msg_id", which is stored
823 in the message envelope with the sound files. IMAP backends will now store
824 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
825 backends will store the message identifier in a "msg_id" column. See
826 UPGRADE.txt for more information.
828 * Added VoiceMailPlayMsg application. This application will play a single
829 voicemail message from a mailbox. The result of the application, SUCCESS or
830 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
835 * Hangup handlers can be attached to channels using the CHANNEL() function.
836 Hangup handlers will run when the channel is hung up similar to the h
837 extension. The hangup_handler_push option will push a GoSub compatible
838 location in the dialplan onto the channel's hangup handler stack. The
839 hangup_handler_pop option will remove the last added location, and optionally
840 replace it with a new GoSub compatible location. The hangup_handler_wipe
841 option will remove all locations on the stack, and optionally add a new
844 * The expression parser now recognizes the ABS() absolute value function,
845 which will convert negative floating point values to positive values.
847 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
848 control of faxdetect.
850 * Addition of the VM_INFO function that can be used to retrieve voicemail
851 user information, such as the email address and full name.
852 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
855 * The REDIRECTING function now supports the redirecting original party id
858 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
859 lets you set some of the configuration options from the [general] section
860 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
861 the key sequence used to activate built-in features, such as blindxfer,
862 and automon. See the built-in documentation for details.
864 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
865 instead of simply the uri. This is the format that MessageSend() can use
866 in the from parameter for outgoing SIP messages.
868 * Added the PRESENCE_STATE function. This allows retrieving presence state
869 information from any presence state provider. It also allows setting
870 presence state information from a CustomPresence presence state provider.
871 See AMI/CLI changes for related commands.
873 * Added the AMI_CLIENT function to make manager account attributes available
874 to the dialplan. It currently supports returning the current number of
875 active sessions for a given account.
877 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
878 and the REDIRECTING functions.
886 * Added a manager event "LocalBridge" for local channel call bridges between
887 the two pseudo-channels created.
892 * Added dialtone_detect option for analog ports to disconnect incoming
893 calls when dialtone is detected.
895 * Added option colp_send to send ISDN connected line information. Allowed
896 settings are block, to not send any connected line information; connect, to
897 send connected line information on initial connect; and update, to send
898 information on any update during a call. Default is update.
900 * Add options namedcallgroup and namedpickupgroup to support installations
901 where a higher number of groups (>64) is required.
903 * Added support to use private party ID information with PRI calls.
908 * A new channel driver named chan_motif has been added which provides support for
909 Google Talk and Jingle in a single channel driver. This new channel driver includes
910 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
911 hold, unhold, and ringing notification. It is also compliant with the current Jingle
912 specification, current Google Jingle specification, and the original Google Talk
918 * Added NAT support for RTP. Setting in config is 'nat', which can be set
919 globally and overriden on a peer by peer basis.
921 * Direct media functionality has been added. Options in config are:
922 directmedia (directrtp) and directrtpsetup (earlydirect)
924 * ChannelUpdate events now contain a CallRef header.
929 * Asterisk will no longer substitute CID number for CID name in the display
930 name field if CID number exists without a CID name. This change improves
931 compatibility with certain device features such as Avaya IP500's directory
934 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
935 created using that setting to not be removed during SIP reload.
937 * Added settings recordonfeature and recordofffeature. When receiving an INFO
938 request with a "Record:" header, this will turn the requested feature on/off.
939 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
940 dynamic features must be enabled and configured properly on the requesting
941 channel for this to function properly.
943 * Add support to realtime for the 'callbackextension' option.
945 * When multiple peers exist with the same address, but differing
946 callbackextension options, incoming requests that are matched by address
947 will be matched to the peer with the matching callbackextension if it is
950 * Two new NAT options, auto_force_rport and auto_comedia, have been added
951 which set the force_rport and comedia options automatically if Asterisk
952 detects that an incoming SIP request crossed a NAT after being sent by
955 * The default global nat setting in sip.conf has been changed from force_rport
958 * NAT settings are now a combinable list of options. The equivalent of the
959 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
961 * Adds an option send_diversion which can be disabled to prevent
962 diversion headers from automatically being added to INVITE requests.
964 * Add support for lightweight NAT keepalive. If enabled a blank packet will
965 be sent to the remote host at a given interval to keep the NAT mapping open.
966 This can be enabled using the keepalive configuration option.
968 * Add option 'tonezone' to specify country code for indications. This option
969 can be set both globally and overridden for specific peers.
971 * The SIP Security Events Framework now supports IPv6.
973 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
974 between multiple user agents. When set, for directmedia reinvites,
975 Asterisk will not send an immediate reinvite on an incoming call leg. This
976 option is useful when peered with another SIP user agent that is known to
977 send immediate direct media reinvites upon call establishment.
979 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
982 * Add options subminexpiry and submaxexpiry to set limits of subscription
983 timer independently from registration timer settings. The setting of the
984 registration timer limits still is done by options minexpiry, maxexpiry
985 and defaultexpiry. For backwards compatibility the setting of minexpiry
986 and maxexpiry also is used to configure the subscription timer limits if
987 subminexpiry and submaxexpiry are not set in sip.conf.
989 * Set registration timer limits to default values when reloading sip
990 configuration and values are not set by configuration.
992 * Add options namedcallgroup and namedpickupgroup to support installations
993 where a higher number of groups (>64) is required.
995 * When a MESSAGE request is received, the address the request was received from
996 is now saved in the SIP_RECVADDR variable.
998 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
999 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
1000 the ANI2/OLI information is set on the channel, which can be retrieved using
1001 the CALLERID function.
1003 * Peers can now be configured to support negotiation of ICE candidates using
1004 the setting icesupport. See res_rtp_asterisk changes for more information.
1006 * Added support for format attribute negotiation. See the Codecs changes for
1009 * Extra headers specified with SIPAddHeader are sent with the REFER message
1010 when using Transfer application. See refer_addheaders in sip.conf.sample.
1012 * Added support to use private party ID information with calls.
1014 * Adds an option discard_remote_hold_retrieval that when set stops telling
1015 the peer to start music on hold.
1020 * Added skinny version 17 protocol support.
1024 --------------------
1025 * Added ability to use multiple lines for a single phone. This allows multiple
1026 calls to occur on a single phone, using callwaiting and switching between calls.
1028 * Added option 'sharpdial' allowing end dialing by pressing # key
1030 * Added option 'interdigit_timer' to control phone dial timeout
1032 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
1034 * Added global 'debug' option, that enables debug in channel driver
1036 * Added ability to translate on-screen menu in multiple languages. Tested on
1037 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
1038 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
1041 * In addition to English added French and Russian languages for on-screen menus
1043 * Reworked dialing number input: added dialing by timeout, immediate dial on
1044 on dialplan compare, phone number length now not limited by screen size
1046 * Added ability to pickup a call using features.conf defined value and
1052 * Add options namedcallgroup and namedpickupgroup to support installations
1053 where a higher number of groups (>64) is required.
1055 * Added support to use private party ID information with calls.
1060 * The minimum DTMF duration can now be configured in asterisk.conf
1061 as "mindtmfduration". The default value is (as before) set to 80 ms.
1062 (previously it was only available in source code)
1064 * Named ACLs can now be specified in acl.conf and used in configurations that
1065 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
1066 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
1067 working ACL. In addition, some CLI commands have been added to provide
1068 show information and allow for module reloading - see CLI Changes.
1070 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
1071 items (separated by commas), and items in the rule can be negated by prefixing
1072 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
1073 longer necessray to control the order that the 'permit' and 'deny' columns are
1074 returned from queries.
1076 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
1077 be used within the dynamic weight attribute when specifying a mapping.
1079 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
1080 header, instead of putting the user defined event name there. When enabled
1081 the UserDefType header is added for user defined events. This feature is
1082 enabled with the setting show_user_defined.
1084 * Macro has been deprecated in favor of GoSub. For redirecting and connected
1085 line purposes use the following variables instead of their macro equivalents:
1086 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
1087 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
1088 cc_callback_macro in channel configurations.
1090 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
1093 * Call files now support the "early_media" option to connect with an outgoing
1094 extension when early media is received.
1096 * Added support to use private party ID information with calls.
1101 * A new channel variable, AGIEXITONHANGUP, has been added which allows
1102 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
1103 AGI application would exit immediately after a channel hangup is detected.
1105 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
1106 are resolved and each address is attempted in turn until one succeeds or
1110 AMI (Asterisk Manager Interface)
1112 * The originate action now has an option "EarlyMedia" that enables the
1113 call to bridge when we get early media in the call. Previously,
1114 early media was disregarded always when originating calls using AMI.
1116 * Added setvar= option to manager accounts (much like sip.conf)
1118 * Originate now generates an error response if the extension given is not found
1121 * MixMonitor will now show IDs associated with the mixmonitor upon creating
1122 them if the i(variable) option is used. StopMixMonitor will accept
1123 MixMonitorID as an option to close specific MixMonitors.
1125 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
1126 updated to include information about peers configured with
1127 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
1128 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
1129 returned if auto_force_rport is not enabled.
1131 * Added SIPpeerstatus manager command which will generate PeerStatus events
1132 similar to the existing PeerStatus events found in chan_sip on demand.
1134 * Hangup now can take a regular expression as the Channel option. If you want
1135 to hangup multiple channels, use /regex/ as the Channel option. Existing
1136 behavior to hanging up a single channel is unchanged, but if you pass a regex,
1137 the manager will send you a list of channels back that were hung up.
1139 * Support for IPv6 addresses has been added.
1141 * AMI Events can now be documented in the Asterisk source. Note that AMI event
1142 documentation is only generated when Asterisk is compiled using 'make full'.
1143 See the CLI section for commands to display AMI event information.
1145 * The AMI Hangup event now includes the AccountCode header so you can easily
1146 correlate with AMI Newchannel events.
1148 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
1149 the StateInterface of the queue member.
1151 * Added AMI event SessionTimeout in the Call category that is issued when a
1152 call is terminated due to either RTP stream inactivity or SIP session timer
1155 * CEL events can now contain a user defined header UserDefType. See core
1156 changes for more information.
1158 * OOH323 ChannelUpdate events now contain a CallRef header.
1160 * Added PresenceState command. This command will report the presence state for
1161 the given presence provider.
1163 * Added Parkinglots command. This will list all parking lots as a series of
1164 AMI Parkinglot events.
1166 * Added MessageSend command. This behaves in the same manner as the
1167 MessageSend application, and is a technolgoy agnostic mechanism to send out
1168 of call text messages.
1170 * Added "message" class authorization. This grants an account permission to
1171 send out of call messages. Write-only.
1176 * The "dialplan add include" command has been modified to create context a context
1177 if one does not already exist. For instance, "dialplan add include foo into bar"
1178 will create context "bar" if it does not already exist.
1180 * A "dialplan remove context" command has been added to remove a context from
1183 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
1184 filenames of all running mixmonitors on a channel.
1186 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
1187 numeric instead of 0, 1, or 2.
1189 * "stun show status" will show a table describing how the STUN client is
1192 * "acl show [named acl]" will show information regarding a Named ACL. The
1193 acl module can be reloaded with "reload acl".
1195 * Added CLI command to display AMI event information - "manager show events",
1196 which shows a list of all known and documented AMI events, and "manager show
1197 event [event name]", which shows detail information about a specific AMI
1200 * The result of the CLI command "queue show" now includes the state interface
1201 information of the queue member.
1203 * The command "core set verbose" will now set a separate level of logging for
1204 each remote console without affecting any other console.
1206 * Added command "cdr show pgsql status" to check connection status
1208 * "sip show channel" will now display the complete route set.
1210 * Added "presencestate list" command. This command will list all custom
1211 presence states that have been set by using the PRESENCE_STATE dialplan
1214 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
1215 command. This changes a custom presence to a new state.
1220 * Codec lists may now be modified by the '!' character, to allow succinct
1221 specification of a list of codecs allowed and disallowed, without the
1222 requirement to use two different keywords. For example, to specify all
1223 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
1225 * Add support for parsing SDP attributes, generating SDP attributes, and
1226 passing it through. This support includes codecs such as H.263, H.264, SILK,
1227 and CELT. You are able to set up a call and have attribute information pass.
1228 This should help considerably with video calls.
1230 * The iLBC codec can now use a system-provided iLBC library if one is installed,
1231 just like the GSM codec.
1235 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
1236 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
1240 * Asterisk version and build information is now logged at the beginning of a
1243 * Threads belonging to a particular call are now linked with callids which get
1244 added to any log messages produced by those threads. Log messages can now be
1245 easily identified as involved with a certain call by looking at their call id.
1246 Call ids may also be attached to log messages for just about any case where
1247 it can be determined to be related to a particular call.
1249 * Each logging destination and console now have an independent notion of the
1250 current verbosity level. Logger.conf now allows an optional argument to
1251 the 'verbose' specifier, indicating the level of verbosity sent to that
1252 particular logging destination. Additionally, remote consoles now each
1253 have their own verbosity level. The command 'core set verbose' will now set
1254 a separate level for each remote console without affecting any other
1260 * Added 'announcement' option which will play at the start of MOH and between
1261 songs in modes of MOH that can detect transitions between songs (eg.
1267 * New per parking lot options: comebackcontext and comebackdialtime. See
1268 configs/features.conf.sample for more details.
1270 * Channel variable PARKER is now set when comebacktoorigin is disabled in
1273 * Channel variable PARKEDCALL is now set with the name of the parking lot
1274 when a timeout occurs.
1280 CDR Postgresql Driver
1282 * Added command "cdr show pgsql status" to check connection status
1285 CDR Adaptive ODBC Driver
1287 * Added schema option for databases that support specifying a schema.
1295 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
1296 CALENDAR_WRITE has completed successfully.
1301 * A new option, 'probation' has been added to rtp.conf
1302 RTP in strictrtp mode can now require more than 1 packet to exit learning
1303 mode with a new source (and by default requires 4). The probation option
1304 allows the user to change the required number of packets in sequence to any
1305 desired value. Use a value of 1 to essentially restore the old behavior.
1306 Also, with strictrtp on, Asterisk will now drop all packets until learning
1307 mode has successfully exited. These changes are based on how pjmedia handles
1308 media sources and source changes.
1310 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
1311 enabled or disabled using the icesupport setting. A variety of other
1312 settings have been introduced to configure STUN/TURN connections.
1317 * A new module, res_corosync, has been introduced. This module uses the
1318 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
1319 of Asterisk servers to both Message Waiting Indication (MWI) and/or
1320 Device State (presence) information. This module is very similar to, and
1321 is a replacement for the res_ais module that was in previous releases of
1327 * This module adds a cleaned up, drop-in replacement for res_jabber called
1328 res_xmpp. This provides the same externally facing functionality but is
1329 implemented differently internally. res_jabber has been deprecated in favor
1330 of res_xmpp; please see the UPGRADE.txt file for more information.
1335 * The safe_asterisk script has been updated to allow several of its parameters
1336 to be set from environment variables. This also enables a custom run
1337 directory of Asterisk to be specified, instead of defaulting to /tmp.
1339 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
1340 its value to determine the directory to assume is the top-level directory of
1341 the source tree. If the variable is not set, it defaults to the current
1342 behavior and uses the current working directory.
1344 ------------------------------------------------------------------------------
1345 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
1346 ------------------------------------------------------------------------------
1350 * Asterisk now has protocol independent support for processing text messages
1351 outside of a call. Messages are routed through the Asterisk dialplan.
1352 SIP MESSAGE and XMPP are currently supported. There are options in
1353 jabber.conf and sip.conf to allow enabling these features.
1354 -> jabber.conf: see the "sendtodialplan" and "context" options.
1355 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
1356 and "outofcall_message_context" options.
1357 The MESSAGE() dialplan function and MessageSend() application have been
1358 added to go along with this functionality. More detailed usage information
1359 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
1360 * If real-time text support (T.140) is negotiated, it will be preferred for
1361 sending text via the SendText application. For example, via SIP, messages
1362 that were once sent via the SIP MESSAGE request would be sent via RTP if
1363 T.140 text is negotiated for a call.
1367 * parkedmusicclass can now be set for non-default parking lots.
1369 Asterisk Manager Interface
1370 --------------------------
1371 * PeerStatus now includes Address and Port.
1372 * Added Hold events for when the remote party puts the call on and off hold
1373 for chan_dahdi ISDN channels.
1374 * Added new action MeetmeListRooms to list active conferences (shows same
1375 data as "meetme list" at the CLI).
1376 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
1377 Description field that is set by 'description' in the channel configuration
1379 * Added Uniqueid header to UserEvent.
1380 * Added new action FilterAdd to control event filters for the current session.
1381 This requires the system permission and uses the same filter syntax as
1382 filters that can be defined in manager.conf
1383 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
1384 versions had some instances of the event converted, but others were left
1385 as-is. All Unlink events should now be converted to Bridge events. The AMI
1386 protocol version number was incremented to 1.2 as a result of this change.
1388 Asterisk HTTP Server
1389 --------------------------
1390 * The HTTP Server can bind to IPv6 addresses.
1393 --------------------------
1394 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
1395 with busydetect. usage example: busypattern=200,200,200,600
1398 --------------------------
1399 * New 'gtalk show settings' command showing the current settings loaded from
1401 * The 'logger reload' command now supports an optional argument, specifying an
1402 alternate configuration file to use.
1403 * 'dialplan add extension' command will now automatically create a context if
1404 the specified context does not exist with a message indicated it did so.
1405 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
1406 Description field which can be populated with 'description' in the channel
1407 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
1410 --------------------------
1411 * The filter option in cdr_adaptive_odbc now supports negating the argument,
1412 thus allowing records which do NOT match the specified filter.
1413 * Added ability to log CONGESTION calls to CDR
1416 --------------------------
1417 * Ability to define custom SILK formats in codecs.conf.
1418 * Addition of speex32 audio format with translation.
1419 * CELT codec pass-through support and ability to define
1420 custom CELT formats in codecs.conf.
1421 * Ability to read raw signed linear files with sample rates
1422 ranging from 8khz - 192khz. The new file extensions introduced
1423 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
1424 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
1425 Skinny, H.323, etc) can still only support the following codecs:
1426 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
1427 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
1428 Video: h261, h263, h263p, h264, mpeg4
1433 --------------------------
1434 * New highly optimized and customizable ConfBridge application capable of
1435 mixing audio at sample rates ranging from 8khz-96khz.
1436 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
1437 and bridge profiles on a channel.
1438 * CONFBRIDGE_INFO dialplan function capable of retrieving information
1439 about a conference such as locked status and number of parties, admins,
1441 * Addition of video_mode option in confbridge.conf for adding video support
1442 into a bridge profile.
1443 * Addition of the follow_talker video_mode in confbridge.conf. This video
1444 mode dynamically switches the video feed to always display the loudest talker
1445 supplying video in the conference.
1449 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
1450 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
1451 variables from asterisk.conf.
1455 * Addition of the JITTERBUFFER dialplan function. This function allows
1456 for jitterbuffering to occur on the read side of a channel. By using
1457 this function conference applications such as ConfBridge and MeetMe can
1458 have the rx streams jitterbuffered before conference mixing occurs.
1459 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
1461 * Added STRREPLACE function. This function let's the user search a variable
1462 for a given string to replace with another string as many times as the
1463 user specifies or just throughout the whole string.
1464 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
1465 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
1466 * Added extensions to chan_ooh323 in function CHANNEL()
1468 libpri channel driver (chan_dahdi) DAHDI changes
1469 --------------------------
1470 * Added moh_signaling option to specify what to do when the channel's bridged
1471 peer puts the ISDN channel on hold.
1472 * Added display_send and display_receive options to control how the display ie
1473 is handled. To send display text from the dialplan use the SendText()
1474 application when the option is enabled.
1475 * Added mcid_send option to allow sending a MCID request on a span.
1478 --------------------------
1479 * Added setvar option to calendar.conf to allow setting channel variables on
1480 notification channels.
1481 * Added "calendar show types" CLI command to list registered calendar
1485 --------------------------
1486 * Added two new options, r and t with file name arguments to record
1487 single direction (unmixed) audio recording separate from the bidirectional
1488 (mixed) recording. The mixed file name argument is optional now as long
1489 as at least one recording option is used.
1492 --------------------------
1493 * Added a new option, l, which will disable local call optimization for
1494 channels involved with the FollowMe thread. Use this option to improve
1495 compatability for a FollowMe call with certain dialplan apps, options, and
1499 --------------------------
1500 * Added option "k" that will automatically close the conference when there's
1501 only one person left when a user exits the conference.
1504 --------------------------
1505 * cel_pgsql now supports the 'extra' column for data added using the
1506 CELGenUserEvent() application.
1509 --------------------------
1510 * Support for defining hints has been added to pbx_lua. See the 'hints' table
1511 in the sample extensions.lua file for syntax details.
1512 * Applications that perform jumps in the dialplan such as Goto will now
1513 execute properly. When pbx_lua detects that the context, extension, or
1514 priority we are executing on has changed it will immediately return control
1515 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
1516 the priority after the currently executing priority.
1517 * An autoservice is now started by default for pbx_lua channels. It can be
1518 stopped and restarted using the autoservice_stop() and autoservice_start()
1522 --------------------------
1523 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
1524 into a FAXStatus event with an 'Operation' header that will be either
1525 'send', 'receive', and 'gateway'.
1526 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
1527 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
1528 feature will handle converting a fax call between an audio T.30 fax terminal
1529 and an IFP T.38 fax terminal.
1533 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
1534 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
1535 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
1539 * Added general option negative_penalty_invalid default off. when set
1540 members are seen as invalid/logged out when there penalty is negative.
1541 for realtime members when set remove from queue will set penalty to -1.
1542 * Added queue option autopausedelay when autopause is enabled it will be
1543 delayed for this number of seconds since last successful call if there
1544 was no prior call the agent will be autopaused immediately.
1545 * Added member option ignorebusy this when set and ringinuse is not
1546 will allow per member control of multiple calls as ringinuse does for
1551 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
1553 * Added 'k' option to MeetMe to automatically kill the conference when there's only
1554 one participant left (much like a normal call bridge)
1555 * Added extra argument to Originate to set timeout.
1559 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
1560 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
1561 utility in the UTILS section of menuselect. If an existing astdb is found and no
1562 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
1563 convert an existing astdb to the SQLite3 version automatically at runtime.
1567 * Modules marked as deprecated are no longer marked as building by default. Enabling
1568 these modules is still available via menuselect.
1572 * authdebug is now disabled by default. To enable this functionaility again
1573 set authdebug = yes in iax.conf.
1577 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
1578 releases it was disabled.
1582 * The PBX core previously made a call with a non-existing extension test for
1583 extension s@default and jump there if the extension existed.
1584 This was a bad default behaviour and violated the principle of least surprise.
1585 It has therefore been changed in this release. It may affect some
1586 applications and configurations that rely on this behaviour. Most channel
1587 drivers have avoided this for many releases by testing whether the extension
1588 called exists before starting the PBX and generating a local error.
1589 This behaviour still exists and works as before.
1591 Extension "s" is used when no extension is given in a channel driver,
1592 like immediate answer in DAHDI or calling to a domain with no user part
1595 ------------------------------------------------------------------------------
1596 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
1597 ------------------------------------------------------------------------------
1601 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
1602 now defaults to force_rport. It is very important that phones requiring nat=no be
1603 specifically set as such instead of relying on the default setting. If at all
1604 possible, all devices should have nat settings configured in the general section as
1605 opposed to configuring nat per-device.
1606 * Added preferred_codec_only option in sip.conf. This feature limits the joint
1607 codecs sent in response to an INVITE to the single most preferred codec.
1608 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
1609 to be used for the outgoing call. It must be one of the codecs configured
1611 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
1612 to be used for holding a private key. If tlsprivatekey is not specified,
1613 tlscertfile is searched for both public and private key.
1614 * Added tlsclientmethod option to sip.conf. This allows the protocol for
1615 outbound client connections to be specified.
1616 * The sendrpid parameter has been expanded to include the options
1617 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
1618 header to be sent (equivalent to setting sendrpid=yes) and setting
1619 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
1620 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
1621 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
1622 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
1623 will accept the SDP even if the SDP version number is not properly incremented,
1624 but will generate a warning in the log indicating that the SIP peer that sent
1625 the SDP should have the 'ignoresdpversion' option set.
1626 * The 'nat' option has now been been changed to have yes, no, force_rport, and
1627 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
1628 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
1629 remote side requests it and disables symmetric RTP support. Setting it to
1630 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
1631 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
1632 and enables symmetric RTP support.
1633 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
1634 response. This permits the master channel to know how each channel dialled
1635 in a multi-channel setup resolved in an individual way. This carries a
1636 performance penalty and can be disabled in sip.conf using the
1637 'storesipcause' option.
1638 * Added 'externtcpport' and 'externtlsport' options to allow custom port
1639 configuration for the externip and externhost options when tcp or tls is used.
1640 * Added support for message body (stored in content variable) to SIP NOTIFY message
1641 accessible via AMI and CLI.
1642 * Added 'media_address' configuration option which can be used to explicitly specify
1643 the IP address to use in the SDP for media (audio, video, and text) streams.
1644 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
1645 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
1647 * Added 'use_q850_reason' configuration option for generating and parsing
1648 if available Reason: Q.850;cause=<cause code> header. It is implemented
1649 in some gateways for better passing PRI/SS7 cause codes via SIP.
1650 * When dialing SIP peers, a new component may be added to the end of the dialstring
1651 to indicate that a specific remote IP address or host should be used when dialing
1652 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
1653 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
1654 ability to selectively force bridged channels to also be encrypted is also
1655 implemented. Branching in the dialplan can be done based on whether or not
1656 a channel has secure media and/or signaling.
1657 * Added directmediapermit/directmediadeny to limit which peers can send direct media
1659 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
1660 Charge messages to snom phones.
1661 * Added support for G.719 media streams.
1662 * Added support for 16khz signed linear media streams.
1663 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
1664 RTP has been outfitted with the same abilities.
1665 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
1666 available in device configurations as well as in the dial plan.
1667 * Addition of the 'subscribe_network_change' option for turning on and off
1668 res_stun_monitor module support in chan_sip.
1669 * Addition of the 'auth_options_requests' option for turning on and off
1670 authentication for OPTIONS requests in chan_sip.
1674 * Add #tryinclude statement for config files. This provides the same
1675 functionality as the #include statement however an asterisk module will
1676 still load if the filename does not exist. Using the #include statement
1677 Asterisk will not allow the module to load.
1681 * Added rtsavesysname option into iax.conf to allow the systname to be saved
1682 on realtime updates.
1683 * Added the ability for chan_iax2 to inform the dialplan whether or not
1684 encryption is being used. This interoperates with the SIP SRTP implementation
1685 so that a secure SIP call can be bridged to a secure IAX call when the
1686 dialplan requires bridged channels to be "secure".
1687 * Addition of the 'subscribe_network_change' option for turning on and off
1688 res_stun_monitor module support in chan_iax.
1693 * Added ability to preset channel variables on indicated lines with the setvar
1694 configuration option. Also, clearvars=all resets the list of variables back
1696 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
1697 See configs/res_pktccops.conf for more information.
1699 XMPP Google Talk/Jingle changes
1700 -------------------------------
1701 * Added the externip option to gtalk.conf.
1702 * Added the stunaddr option to gtalk.conf which allows for the automatic
1703 retrieval of the external ip from a stun server.
1707 * Added 'p' option to PickupChan() to allow for picking up channel by the first
1708 match to a partial channel name.
1709 * Added .m3u support for Mp3Player application.
1710 * Added progress option to the app_dial D() option. When progress DTMF is
1711 present, those values are sent immediately upon receiving a PROGRESS message
1712 regardless if the call has been answered or not.
1713 * Added functionality to the app_dial F() option to continue with execution
1714 at the current location when no parameters are provided.
1715 * Added the 'a' option to app_dial to answer the calling channel before any
1716 announcements or macros are executed.
1717 * Modified app_dial to set answertime when the called channel answers even if
1718 the called channel hangs up during playback of an announcement.
1719 * Modified app_dial 'r' option to support an additional parameter to play an
1720 indication tone from indications.conf
1721 * Added c() option to app_chanspy. This option allows custom DTMF to be set
1722 to cycle through the next available channel. By default this is still '*'.
1723 * Added x() option to app_chanspy. This option allows DTMF to be set to
1724 exit the application.
1725 * The Voicemail application has been improved to automatically ignore messages
1726 that only contain silence.
1727 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
1728 associated mailbox(es) to be greetings-only.
1729 * The ChanSpy application now has the 'S' option, which makes the application
1730 automatically exit once it hits a point where no more channels are available
1732 * The ChanSpy application also now has the 'E' option, which spies on a single
1733 channel and exits when that channel hangs up.
1734 * The MeetMe application now turns on the DENOISE() function by default, for
1735 each participant. In our tests, this has significantly decreased background
1736 noise (especially noisy data centers).
1737 * Voicemail now permits storage of secrets in a separate file, located in the
1738 spool directory of each individual user. The control for this is located in
1739 the "passwordlocation" option in voicemail.conf. Please see the sample
1740 configuration for more information.
1741 * The ChanIsAvail application now exposes the returned cause code using a separate
1742 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
1743 * Added 'd' option to app_followme. This option disables the "Please hold"
1745 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
1746 received will terminate recording.
1747 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
1748 Previously the folder could only be set per context, but has now been extended
1749 using the imapfolder option.
1750 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
1751 * Voicemail now allows the pager date format to be specified separately from the
1753 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
1754 to allow joining, leaving, and sending text to group chats.
1755 * MeetMe has a new option 'G' to play an announcement before joining a conference.
1756 * Page has a new option 'A(x)' which will playback an announcement simultaneously
1757 to all paged phones (and optionally excluding the caller's one using the new
1758 option 'n') before the call is bridged.
1759 * The 'f' option to Dial has been augmented to take an optional argument. If no
1760 argument is provided, the 'f' option works as it always has. If an argument is
1761 provided, then the connected party information of all outgoing channels created
1762 during the Dial will be set to the argument passed to the 'f' option.
1763 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
1765 * The OSP lookup application adds in/outbound network ID, optional security,
1766 number portability, QoS reporting, destination IP port, custom info and service
1768 * Added new application VMSayName that will play the recorded name of the voicemail
1769 user if it exists, otherwise will play the mailbox number.
1770 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
1771 retrieve state for a particular bridge, where <name> is the conference name
1772 * app_directory now allows exiting at any time using the operator or pound key.
1773 * Voicemail now supports setting a locale per-mailbox.
1774 * Two new applications are provided for declining counting phrases in multiple
1775 languages. See the application notes for SayCountedNoun and SayCountedAdj for
1777 * Voicemail now runs the externnotify script when pollmailboxes is activated and
1779 * Voicemail now includes rdnis within msgXXXX.txt file.
1780 * ExternalIVR now supports IPv6 addresses.
1781 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
1782 at https://wiki.asterisk.org/wiki/x/oQBB
1783 * ParkedCall and Park can now specify the parking lot to use.
1787 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
1788 over SRV records associated with a specific service. From the CLI, type
1789 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
1790 details on how these may be used.
1791 * PITCH_SHIFT dialplan function added. This function can be used to modify the
1792 pitch of a channel's tx and rx audio streams.
1793 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
1794 setting various connected line and redirecting party information.
1795 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
1796 support ISDN subaddressing.
1797 * The CHANNEL() function now supports the "name" and "checkhangup" options.
1798 * For DAHDI channels, the CHANNEL() dialplan function now allows
1799 the dialplan to request changes in the configuration of the active
1800 echo canceller on the channel (if any), for the current call only.
1803 exten => s,n,Set(CHANNEL(echocan_mode)=off)
1805 The possible values are:
1807 on - normal mode (the echo canceller is actually reinitialized)
1809 fax - FAX/data mode (NLP disabled if possible, otherwise completely
1811 voice - voice mode (returns from FAX mode, reverting the changes that
1812 were made when FAX mode was requested)
1813 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
1814 and setting variables on the channel which created the current channel.
1815 Administrators should take care to avoid naming conflicts, when multiple
1816 channels are dialled at once, especially when used with the Local channel
1817 construct (which all could set variables on the master channel). Usage
1818 of the HASH() dialplan function, with the key set to the name of the slave
1819 channel, is one approach that will avoid conflicts.
1820 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
1822 * func_odbc now allows multiple row results to be retrieved without using
1823 mode=multirow. If rowlimit is set, then additional rows may be retrieved
1824 from the same query by using the name of the function which retrieved the
1825 first row as an argument to ODBC_FETCH().
1826 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
1827 dialplan. This function returns the content of the received message.
1828 * Added REPLACE, which searches a given variable name for a set of characters,
1829 then either replaces them with a single character or deletes them.
1830 * Added PASSTHRU, which literally passes the same argument back as its return
1831 value. The intent is to be able to use a literal string argument to
1832 functions that currently require a variable name as an argument.
1833 * HASH-associated variables now can be inherited across channel creation, by
1834 prefixing the name of the hash at assignment with the appropriate number of
1835 underscores, just like variables.
1836 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
1837 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
1838 whether or not channels that are bridged to the current channel will be
1839 required to have secure signaling and/or media.
1840 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
1841 the current channel has secure signaling and/or media.
1842 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
1843 "no_media_path" option.
1844 Returns "0" if there is a B channel associated with the call.
1845 Returns "1" if no B channel is associated with the call. The call is either
1846 on hold or is a call waiting call.
1847 * Added option to dialplan function CDR(), the 'f' option
1848 allows for high resolution times for billsec and duration fields.
1849 * FILE() now supports line-mode and writing.
1850 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
1851 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
1855 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
1856 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
1857 and is set when a dynamic feature is triggered.
1858 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
1859 to dynamically create a new parking lot matching the value this varible is
1861 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
1862 features.conf that should be the base for dynamic parkinglots.
1863 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
1864 parkinglot should have.
1865 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
1866 parkinglot should have.
1867 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
1872 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
1873 timeout has expired.
1874 * Added 'R' option to app_queue. This option stops moh and indicates ringing
1875 to the caller when an Agent's phone is ringing. This can be used to indicate
1876 to the caller that their call is about to be picked up, which is nice when
1877 one has been on hold for an extened period of time.
1878 * A new config option, penaltymemberslimit, has been added to queues.conf.
1879 When set this option will disregard penalty settings when a queue has too
1881 * A new option, 'I' has been added to both app_queue and app_dial.
1882 By setting this option, Asterisk will not update the caller with
1883 connected line changes or redirecting party changes when they occur.
1884 * A 'relative-periodic-announce' option has been added to queues.conf. When
1885 enabled, this option will cause periodic announce times to be calculated
1886 from the end of announcements rather than from the beginning.
1887 * The autopause option in queues.conf can be passed a new value, "all." The
1888 result is that if a member becomes auto-paused, he will be paused in all
1889 queues for which he is a member, not just the queue that failed to reach
1891 * Added dialplan function QUEUE_EXISTS to check if a queue exists
1892 * The queue logger now allows events to optionally propagate to a file,
1893 even when realtime logging is turned on. Additionally, realtime logging
1894 supports sending the event arguments to 5 individual fields, although it
1895 will fallback to the previous data definition, if the new table layout is
1898 mISDN channel driver (chan_misdn) changes
1899 ----------------------------------------
1900 * Added display_connected parameter to misdn.conf to put a display string
1901 in the CONNECT message containing the connected name and/or number if
1902 the presentation setting permits it.
1903 * Added display_setup parameter to misdn.conf to put a display string
1904 in the SETUP message containing the caller name and/or number if the
1905 presentation setting permits it.
1906 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
1907 indicate the dialplan settings are to be obtained from the asterisk
1909 * Made misdn.conf parameter callerid accept the "name" <number> format
1910 used by the rest of the system.
1911 * Made use the nationalprefix and internationalprefix misdn.conf
1912 parameters to prefix any received number from the ISDN link if that
1913 number has the corresponding Type-Of-Number. NOTE: This includes
1914 comparing the incoming call's dialed number against the MSN list.
1915 * Added the following new parameters: unknownprefix, netspecificprefix,
1916 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
1917 received number from the ISDN link if that number has the corresponding
1919 * Added new dialplan application misdn_command which permits controlling
1920 the CCBS/CCNR functionality.
1921 * Added new dialplan function mISDN_CC which permits retrieval of various
1922 values from an active call completion record.
1923 * For PTP, you should manually send the COLR of the redirected-to party
1924 for an incomming redirected call if the incoming call could experience
1925 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
1926 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
1927 if the REDIRECTING(from-num) is not empty.
1928 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
1929 option on all of the REDIRECTING statements before dialing the
1930 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
1931 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
1932 redirecting-to presentation (COLR) when it becomes available.
1933 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
1936 thirdparty mISDN enhancements
1937 -----------------------------
1938 mISDN has been modified by Digium, Inc. to greatly expand facility message
1940 * Enhanced COLP support for call diversion and transfer.
1941 * CCBS/CCNR support.
1943 The latest modified mISDN v1.1.x based version is available at:
1944 http://svn.digium.com/svn/thirdparty/mISDN/trunk
1945 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
1947 Tagged versions of the modified mISDN code are available under:
1948 http://svn.digium.com/svn/thirdparty/mISDN/tags
1949 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
1951 libpri channel driver (chan_dahdi) DAHDI changes
1952 -------------------------------------------
1953 * The channel variable PRIREDIRECTREASON is now just a status variable
1954 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
1955 to read and alter the reason.
1956 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
1957 redirected-to party for an incomming redirected call if the incoming call
1958 could experience further redirects. Just set the
1959 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
1960 to the COLR. A call has been redirected if the REDIRECTING(count) is not
1962 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
1963 use the inhibit(i) option on all of the REDIRECTING statements before
1964 dialing the redirected-to party. You still have to set the
1965 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
1966 will update the redirecting-to presentation (COLR) when it becomes available.
1967 * Added the ability to ignore calls that are not in a Multiple Subscriber
1968 Number (MSN) list for PTMP CPE interfaces.
1969 * Added dynamic range compression support for dahdi channels. It is
1970 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
1971 * Added support for ISDN calling and called subaddress with partial support
1972 for connected line subaddress.
1973 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
1974 * Added handling of received HOLD/RETRIEVE messages and the optional ability
1975 to transfer a held call on disconnect similar to an analog phone.
1976 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
1977 Will reroute/deflect an outgoing call when receive the message.
1978 Can use the DAHDISendCallreroutingFacility to send the message for the
1980 * Added standard location to add options to chan_dahdi dialing:
1981 Dial(DAHDI/g1[/extension[/options]])
1984 R Reverse charging indication
1985 * Added Reverse Charging Indication (Collect calls) send/receive option.
1986 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
1987 Dial(DAHDI/g1/extension/R)
1988 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
1989 (requires latest LibPRI)
1990 * Added ability to send/receive keypad digits in the SETUP message.
1991 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
1992 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
1993 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
1994 (requires latest LibPRI)
1995 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
1996 to eliminate tromboned calls. A tromboned call goes out an interface and comes
1997 back into the same interface. Tromboned calls happen because of call routing,
1998 call deflection, call forwarding, and call transfer.
1999 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
2000 * Added the ability to support call waiting calls. (The SETUP has no B channel
2002 * Added Malicious Call ID (MCID) event to the AMI call event class.
2003 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
2005 Asterisk Manager Interface
2006 --------------------------
2007 * The Hangup action now accepts a Cause header which may be used to
2008 set the channel's hangup cause.
2009 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
2010 to specify a separate .pem file to hold a private key. By default sslcert
2011 is used to hold both the public and private key.
2012 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
2013 for options containing the 'tls' prefix. For example, 'sslenable' is now
2014 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
2015 across all .conf files. All affected sample.conf files have been modified to
2016 reflect this change. Previous options such as 'sslenable' still work,
2017 but options with the 'tls' prefix are preferred.
2018 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
2019 in a channel. (res_mutestream.so)
2020 * The configuration file manager.conf now supports a channelvars option, which
2021 specifies a list of channel variables to include in each channel-oriented
2023 * The redirect command now has new parameters ExtraContext, ExtraExtension,
2024 and ExtraPriority to allow redirecting the second channel to a different
2025 location than the first.
2026 * Added new event "JabberStatus" in the Jabber module to monitor buddies
2028 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
2029 in a MixMonitor recording.
2030 * The 'iax2 show peers' output is now similar to the expected output of
2032 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
2034 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
2035 AOC-E messages on a channel.
2036 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
2037 conform more closely to similar events.
2038 * Added a new eventfilter option per user to allow whitelisting and blacklisting
2040 * Added optional parkinglot variable for park command.
2041 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
2042 if CallerIDNum and CallerIDName headers are also present.
2044 Channel Event Logging
2045 ---------------------
2046 * A new interface, CEL, is introduced here. CEL logs single events, much like
2047 the AMI, but it differs from the AMI in that it logs to db backends much
2048 like CDR does; is based on the event subsystem introduced by Russell, and
2049 can share in all its benefits; allows multiple backends to operate like CDR;
2050 is specialized to event data that would be of concern to billing sytems,
2051 like CDR. Backends for logging and accounting calls have been produced,
2052 but a new CDR backend is still in development.
2056 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
2057 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
2058 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
2059 * Multiple files and formats can now be specified in cdr_custom.conf.
2060 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
2061 See configs/cdr_syslog.conf.sample for more information.
2062 * A 'sequence' field has been added to CDRs which can be combined with
2063 linkedid or uniqueid to uniquely identify a CDR.
2064 * Handling of billsec and duration field has changed. If your table definition
2065 specifies those fields as float,double or similar they will now be logged with
2066 microsecond accuracy instead of a whole integer.
2068 Calendaring for Asterisk
2069 ------------------------
2070 * A new set of modules were added supporing calendar integration with Asterisk.
2071 Dialplan functions for reading from and writing to calendars are included,
2072 as well as the ability to execute dialplan logic upon calendar event notifications.
2073 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
2074 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
2075 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
2076 2003 support does not support forms-based authentication).
2078 Call Completion Supplementary Services for Asterisk
2079 ---------------------------------------------------
2080 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
2081 DAHDI/ISDN supports call completion for the following switch types:
2082 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
2083 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
2085 Multicast RTP Support
2086 ---------------------
2087 * A new RTP engine and channel driver have been added which supports Multicast RTP.
2088 The channel driver can be used with the Page application to perform multicast RTP
2089 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
2090 Type can be either basic or linksys.
2091 Destination is the IP address and port for the RTP packets.
2092 Control address is specific to the linksys type and is used for sending the control
2093 packets unique to them.
2095 Security Events Framework
2096 -------------------------
2097 * Asterisk has a new C API for reporting security events. The module res_security_log
2098 sends these events to the "security" logger level. Currently, AMI is the only
2099 Asterisk component that reports security events. However, SIP support will be
2100 coming soon. For more information on the security events framework, see the
2101 "Asterisk Security Framework" section of the Asterisk wiki at
2102 https://wiki.asterisk.org/wiki/x/wgBQ
2103 * SIP support was added in Asterisk 10
2104 * This API now supports IPv6 addresses
2108 * A technology independent fax frontend (res_fax) has been added to Asterisk.
2109 * A spandsp based fax backend (res_fax_spandsp) has been added.
2110 * The app_fax module has been deprecated in favor of the res_fax module and
2111 the new res_fax_spandsp backend.
2112 * The SendFAX and ReceiveFAX applications now send their log messages to a
2113 'fax' logger level, instead of to the generic logger levels. To see these
2114 messages, the system's logger.conf file will need to direct the 'fax' logger
2115 level to one or more destinations; the logger.conf.sample file includes an
2116 example of how to do this. Note that if the 'fax' logger level is *not*
2117 directed to at least one destination, log messages generated by these
2118 applications will be lost, and that if the 'fax' logger level is directed to
2119 the console, the 'core set verbose' and 'core set debug' CLI commands will
2120 have no effect on whether the messages appear on the console or not.
2124 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
2125 Now, in order to enable transmitting silence during record the transmit_silence
2126 option should be used. transmit_silence_during_record remains a valid option, but
2127 defaults to the behavior of the transmit_silence option.
2128 * Addition of the Unit Test Framework API for managing registration and execution
2129 of unit tests with the purpose of verifying the operation of C functions.
2130 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
2131 XMPP text messages to the remote JID.
2132 * Modules.conf has a new option - "require" - that marks a module as critical for
2133 the execution of Asterisk.
2134 If one of the required modules fail to load, Asterisk will exit with a return
2136 * An 'X' option has been added to the asterisk application which enables #exec support.
2137 This allows #exec to be used in asterisk.conf.
2138 * jabber.conf supports a new option auth_policy that toggles auto user registration.
2139 * A new lockconfdir option has been added to asterisk.conf to protect the
2140 configuration directory (/etc/asterisk by default) during reloads.
2141 * The parkeddynamic option has been added to features.conf to enable the creation
2142 of dynamic parkinglots.
2143 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
2144 the reportalarms config option.
2145 * chan_dahdi supports dialing configuring and dialing by device file name.
2146 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
2147 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
2148 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
2149 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
2150 Handy for the above name-based syntax as it does not depend on
2151 initialization order.
2152 * The Realtime dialplan switch now caches entries for 1 second. This provides a
2153 significant increase in performance (about 3X) for installations using this switchtype.
2154 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
2155 AIS. For more information, please see the Distributed Device State section of the
2156 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
2157 * The addition of G.719 pass-through support.
2158 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
2159 during device configuration.
2160 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
2161 have less than 3 lines on the LCD.
2162 * Realtime now supports database failover. See the sample extconfig.conf for details.
2163 * The addition of improved translation path building for wideband codecs. Sample
2164 rate changes during translation are now avoided unless absolutely necessary.
2165 * The addition of the res_stun_monitor module for monitoring and reacting to network
2166 changes while behind a NAT.
2167 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
2168 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
2169 These allow support for any Administration. Default is AT&T values.
2173 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
2174 optionally accept a filename, to apply the setting only to the code generated from
2175 that source file when Asterisk was built. However, there are some modules in Asterisk
2176 that are composed of multiple source files, so this did not result in the behavior
2177 that users expected. In this version, 'core set debug' and 'core set verbose'
2178 can optionally accept *module* names instead (with or without the .so extension),
2179 which applies the setting to the entire module specified, regardless of which source
2180 files it was built from.
2181 * New 'manager show settings' command showing the current settings loaded from
2183 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
2184 the channel hangup request to all channels.
2185 * Added a "core reload" CLI command that executes a global reload of Asterisk.
2187 ------------------------------------------------------------------------------
2188 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
2189 ------------------------------------------------------------------------------
2193 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
2194 Snom phones use this for call pickup of extensions that the phone is
2196 * Added support for setting the domain in the URI for caller of an
2197 outbound call by using the SIPFROMDOMAIN channel variable.
2198 * Added a new configuration option "remotesecret" for authentication to
2199 remote services. For backwards compatibility, "secret" still has the
2200 same function as before, but now you can configure both a remote secret and a
2201 local secret for mutual authentication.
2202 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
2203 the sound will be played to the target of an attended transfer
2204 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
2205 finer control over how many peers Asterisk will qualify and the gap between them
2206 when all peers need to be qualified at the same time.
2207 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
2208 (either globally or for a specific peer), chan_sip will treat any SDP data
2209 it receives as new data and update the media stream accordingly. By
2210 default, Asterisk will only modify the media stream if the SDP session
2211 version received is different from the current SDP session version. This
2212 option is required to interoperate with devices that have non-standard SDP
2213 session version implementations (observed with Microsoft OCS). This option
2214 is disabled by default.
2215 * The parsing of register => lines in sip.conf has been modified to allow a port
2216 to be present in the "user" portion. Please see the sip.conf.sample file for more
2218 * Added support for subscribing to MWI on a remote server and making the status available
2219 as a mailbox. Please see the sip.conf.sample file for more information.
2220 * Added a function to remove SIP headers added in the dialplan before the
2221 first INVITE is generated - SIPRemoveHeader()
2222 * Channel variables set with setvar= in a device configuration is now
2223 set both for inbound and outbound calls.
2224 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
2228 * Added immediate option to iax.conf
2229 * Added forceencryption option to iax.conf
2230 * Added Encryption and Trunk status to manager command "iaxpeers"
2234 * The configuration file now holds separate sections for devices and lines.
2235 Please have a look at configs/skinny.conf.sample and change your skinny.conf
2240 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
2241 support for LibOpenR2. http://www.libopenr2.org/
2242 * The UK option waitfordialtone has been added for use with BT analog
2244 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
2245 is used in conjunction with the 'faxdetect' configuration option. When
2246 'faxbuffers' is used and fax tones are detected, the channel will dynamically
2247 switch to the configured faxbuffers policy. For example, to use 6 buffers
2248 and a 'full' buffer policy for a fax transmission, add:
2250 The faxbuffers configuration will be in affect until the call is torn down.
2251 * Added service message support for 4ESS/5ESS switches.
2255 * For DAHDI channels, the CHANNEL() dialplan function now
2256 supports changing the channel's buffer policy (for the current
2257 call only), using this syntax:
2259 exten => s,n,Set(CHANNEL(buffers)=6,full)
2261 This would change the channel to the 'full' buffer policy and
2262 6 (six) buffers. Possible options for this setting are the same
2263 as those in chan_dahdi.conf.
2264 * Added a new dialplan function, CURLOPT, which permits setting various
2265 options that may be useful with the CURL dialplan function, such as
2266 cookies, proxies, connection timeouts, passwords, etc.
2267 * Permit the syntax and synopsis fields of the corresponding dialplan
2268 functions to be individually set from func_odbc.conf.
2269 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
2270 * func_odbc now may specify an insert query to execute, when the write query
2271 affects 0 rows (usually indicating that no such row exists).
2272 * Added a new dialplan function, LISTFILTER, which permits removing elements
2273 from a set list, by name. Uses the same general syntax as the existing CUT
2274 and FIELDQTY dialplan functions, which also manage lists.
2275 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
2276 obtaining realtime data from the dialplan.
2277 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
2278 a subroutine when using the GoSub() and Return() applications.
2279 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
2280 of "core show function AUDIOHOOK_INHERIT" from the CLI
2281 * Added AES_ENCRYPT. For information on its use, please see the output
2282 of "core show function AES_ENCRYPT" from the CLI
2283 * Added AES_DECRYPT. For information on its use, please see the output
2284 of "core show function AES_DECRYPT" from the CLI
2285 * func_odbc now supports database transactions across multiple queries.
2289 * Scheduled meetme conferences may now have their end times extended by
2291 * app_authenticate now gives the ability to select a prompt other than
2293 * app_directory now pays attention to the searchcontexts setting in
2294 voicemail.conf and will look through all contexts, if no context is
2295 specified in the initial argument.
2296 * A new application, Originate, has been introduced, that allows asynchronous
2297 call origination from the dialplan.
2298 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
2299 in addition to the setting in the "general" context.
2300 * Added ConfBridge dialplan application which does conference bridges without
2301 DAHDI. For information on its use, please see the output of
2302 "core show application ConfBridge" from the CLI.
2306 * The Asterisk CLI has a new command, "channel redirect", which is similar in
2307 operation to the AMI Redirect action.
2308 * extensions.conf now allows you to use keyword "same" to define an extension
2309 without actually specifying an extension. It uses exactly the same pattern
2310 as previously used on the last "exten" line. For example:
2311 exten => 123,1,NoOp(something)
2312 same => n,SomethingElse()
2313 * musiconhold.conf classes of type 'files' can now use relative directory paths,
2314 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
2315 * All deprecated CLI commands are removed from the sourcecode. They are now handled
2316 by the new clialiases module. See cli_aliases.conf.sample file.
2317 * Times within timespecs are now accurate down to the minute. This is a change
2318 from historical Asterisk, which only provided timespecs rounded to the nearest
2319 even (read: evenly divisible by 2) minute mark.
2320 * The realtime switch now supports an option flag, 'p', which disables searches for
2322 * In addition to a time range and date range, timespecs now accept a 5th optional
2323 argument, timezone. This allows you to perform time checks on alternate
2324 timezones, especially if those daylight savings time ranges vary from your
2325 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
2327 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
2328 give you the correct output for an asterisk box behind nat. It will give you the
2329 externhost and localnet settings.
2330 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
2331 can connect calls in passthrough mode, as well as record and play back files.
2332 * Successful and unsuccessful call pickup can now be alerted through sounds, by
2333 using pickupsound and pickupfailsound in features.conf.
2334 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
2335 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
2336 instead of the /var/run/asterisk.pid where it used to be. This will make
2337 installs as non-root easier to manage.
2342 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
2343 be written; they will no longer be explicitly written.
2345 Asterisk Manager Interface
2346 --------------------------
2347 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
2348 a non-empty value) in your request. If you do this, any pending AMI events will
2349 *not* be included in the response to your request as they would normally, but
2350 will be left in the event queue for the next request you make to retrieve. For
2351 some applications, this will allow you to guarantee that you will only see
2352 events in responses to 'WaitEvent' actions, and can better know when to expect them.
2353 To know whether the Asterisk server supports this header or not, your client can
2354 inspect the first response back from the server to see if it includes this header:
2356 Pragma: SuppressEvents
2358 If this is included, the server supports event suppression.
2360 * Added 4 new Actions to list skinny device(s) and line(s)
2366 LDAP Schema File Additions
2367 --------------------------
2368 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
2369 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
2371 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
2372 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
2373 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
2374 * Removed redundant IPaddr (there's already IPAddress)
2375 - Gives more configuration Flags for SIP-Users available (tested)
2376 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
2377 without extensibleObject (which really should be the last resort); gives
2378 also additional possibilities for LDAP-filter
2380 ------------------------------------------------------------------------------
2381 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
2382 ------------------------------------------------------------------------------
2384 Device State Handling
2385 ---------------------
2386 * The event infrastructure in Asterisk got another big update to help support
2387 distributed events. It currently supports distributed device state and
2388 distributed Voicemail MWI (Message Waiting Indication). A new module has
2389 been merged, res_ais, which facilitates communicating events between servers.
2390 It uses the SAForum AIS (Service Availability Forum Application Interface
2391 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
2392 a cluster of Asterisk servers, and to share events between them. For more
2393 information on setting this up, refer to the Distributed Device State section
2394 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
2398 * Added a new dialplan function, AST_CONFIG(), which allows you to access
2399 variables from an Asterisk configuration file.
2400 * The JACK_HOOK function now has a c() option to supply a custom client name.
2401 * Added two new dialplan functions from libspeex for audio gain control and
2402 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
2403 rx directions of a channel from the dialplan.
2404 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
2405 based on other parameters. The default is still to search based on the
2406 forwarding station ID. However, there are new options that allow you to search
2407 based on the message desk terminal ID, or the message desk number.
2408 * TIMEOUT() has been modified to be accurate down to the millisecond.
2409 * ENUM*() functions now include the following new options:
2410 - 'u' returns the full URI and does not strip off the URI-scheme.
2411 - 's' triggers ISN specific rewriting
2412 - 'i' looks for branches into an Infrastructure ENUM tree
2413 - 'd' for a direct DNS lookup without any flipping of digits.
2414 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
2415 * CHANNEL() now has options for the maximum, minimum, and standard or normal
2416 deviation of jitter, rtt, and loss for a call using chan_sip.
2418 DAHDI channel driver (chan_dahdi) Changes
2419 ----------------------------------------
2420 * Channels can now be configured using named sections in chan_dahdi.conf, just
2421 like other channel drivers, including the use of templates.
2422 * The default for pridialplan has changed from 'national' to 'unknown'.
2426 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
2427 to something that matches the pattern a hint will be created using the contents
2428 and variables evaluated.
2429 * Dialplan matching has been extended to allow an extension to return to the
2430 PBX core to wait for more digits. This is done by using the new dialplan
2431 application called "Incomplete". This will permit a whole new level of
2432 extension control, by giving the administrator more control over early
2433 matches employing one of the short-circuit pattern match operators. Note
2434 that custom applications can trigger this same behavior by returning the
2435 special value AST_PBX_INCOMPLETE.
2439 * Directory now permits both first and last names to be matched at the same
2440 time. In addition, the number of digits to enter of the name can be set in
2441 the arguments to Directory; previously, you could enter only 3, regardless
2442 of how many names are in your company. For large companies, this should be
2444 * Voicemail now permits a mailbox setting to wrap around from first to last
2445 messages, if the "messagewrap" option is set to a true value.
2446 * Voicemail now permits an external script to be run, for password validation.
2447 The script should output "VALID" or "INVALID" on stdout, depending upon the
2448 wish to validate or invalidate the password given. Arguments are:
2449 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
2451 * Dial has a new option: F(context^extension^pri), which permits a callee to
2452 continue in the dialplan, at the specified label, if the caller hangs up.
2453 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
2454 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
2455 * The Jack application now has a c() option to supply a custom client name.
2456 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
2457 like the pre-existing whisper mode, except that the spy can also talk to the
2458 participant on the bridged channel as well.
2459 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
2460 to be spoken instead of the channel name or number. For more information on the
2461 use of this option, issue the command "core show application ChanSpy" from the
2463 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
2464 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
2465 words, if using the 'd' option, it is not possible to enter a number to append to
2466 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
2467 change to whisper mode, and pressing 6 will change to barge mode.
2468 * ExternalIVR now takes several options that affect the way it performs, as
2469 well as having several new commands. Please see the External IVR page on the Asterisk
2470 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
2471 * Added ability to communicate over a TCP socket instead of forking a child process for the
2472 ExternalIVR application.
2473 * ChanIsAvail has a new option, 'a', which will return all available channels instead
2474 of just the first one if you give the function more then one channel to check.
2475 * PrivacyManager now takes an option where you can specify a context where the
2476 given number will be matched. This way you have more control over who is allowed
2477 and it stops the people who blindly enter 10 digits.
2478 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
2479 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
2480 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
2481 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
2482 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
2483 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
2484 * The Dial() application no longer copies the language used by the caller to the callee's
2485 channel. If you desire for the caller's channel's language to be used for file playback
2486 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
2487 * SendImage() no longer hangs up the channel on error; instead, it sets the
2488 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
2489 'UNSUPPORTED'. This change makes SendImage() more consistent with other
2491 * Park has a new option, 's', which silences the announcement of the parking space number.
2492 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
2493 invalid input and will be assumed to mean that no timeout is desired.
2497 * Added DNS manager support to registrations for peers referencing peer entries.
2498 DNS manager runs in the background which allows DNS lookups to be run asynchronously
2499 as well as periodically updating the IP address. These properties allow for
2500 better performance as well as recovery in the event of an IP change.
2501 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
2502 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
2503 These changes also provide performance improvements for call setup and tear down.
2504 * Added ability to specify registration expiry time on a per registration basis in
2506 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
2508 * Added t38pt_usertpsource option. See sip.conf.sample for details.
2509 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
2510 * 'sip show peers' and 'sip show users' display their entries sorted in
2511 alphabetical order, as opposed to the order they were in, in the config
2513 * Videosupport now supports an additional option, "always", which always sets
2514 up video RTP ports, even on clients that don't support it. This helps with
2515 callfiles and certain transfers to ensure that if two video phones are
2516 connected, they will always share video feeds.
2520 * Existing DNS manager lookups extended to check for SRV records.
2521 * IAX2 encryption support has been improved to support periodic key rotation
2522 within a call for enhanced security. The option "keyrotate" has been
2523 provided to disable this functionality to preserve backwards compatibility
2524 with older versions of IAX2 that do not support key rotation.
2528 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
2529 data tree based on the given <path>.
2530 * New CLI command "data show providers" that will display all the registered
2532 * New CLI command, "config reload <file.conf>" which reloads any module that
2533 references that particular configuration file. Also added "config list"
2534 which shows which configuration files are in use.
2535 * New CLI commands, "pri show version" and "ss7 show version" that will
2536 display which version of libpri and libss7 are being used, respectively.
2537 A new API call was added so trunk will now have to be compiled against
2538 a versions of libpri and libss7 that have them or it will not know that
2539 these libraries exist.
2540 * The commands "core show globals", "core set global" and "core set chanvar" has
2541 been deprecated in favor of the more semanticly correct "dialplan show globals",
2542 "dialplan set chanvar" and "dialplan set global".
2543 * New CLI command "dialplan show chanvar" to list all variables associated
2544 with a given channel.
2548 * Addresses managed by DNS manager now can check to see if there is a DNS
2549 SRV record for a given domain and will use that hostname/port if present.
2551 AMI - The manager (TCP/TLS/HTTP)
2552 --------------------------------
2553 * The Status command now takes an optional list of variables to display
2554 along with channel status.
2555 * The QueueEntry event now also includes the channel's uniqueid
2559 * res_odbc no longer has a limit of 1023 total possible unshared connections,
2560 as some people were running into this limit. This limit has been increased
2565 * The TRANSFER queue log entry now includes the the caller's original
2566 position in the transferred-from queue.
2567 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
2568 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
2569 as well as an explanation about timeout options in general
2570 * Added a new option - C - for forcing the "answered elsewhere" flag on
2571 cancellation of calls in to members of the queue. This is to avoid the
2572 call to a member of a queue having the call listed as a "missed call".
2576 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
2577 adaptive capabilities. What this means in practical terms is that if your
2578 realtime table lacks critical fields, Asterisk will now emit warnings to
2579 that effect. Also, some of the realtime drivers have the ability (if
2580 configured) to automatically add those columns to the table with the
2581 correct type and length.
2585 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
2586 the 'setvar' option to cause a given audio file to be played upon completion
2587 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
2588 Skinny channels only.
2589 * You can now compile Asterisk against the Hoard Memory Allocator, see the
2590 Hoard page on the Asterisk wiki for more information:
2591 https://wiki.asterisk.org/wiki/x/pQBB
2592 * Config file variables may now be appended to, by using the '+=' append
2593 operator. This is most helpful when working with long SQL queries in
2594 func_odbc.conf, as the queries no longer need to be specified on a single
2596 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
2597 which will add a second to the billsec when the ending
2598 time is set, if the number in the microseconds field of the end time is
2599 greater than the number of microseconds in the answer time. This allows
2600 users to count the 'initiated' seconds in their billing records.
2602 ------------------------------------------------------------------------------
2603 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
2604 ------------------------------------------------------------------------------
2606 AMI - The manager (TCP/TLS/HTTP)
2607 --------------------------------
2608 * Manager has undergone a lot of changes, all of them documented
2609 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
2610 * Manager version has changed to 1.1
2611 * Added a new action 'CoreShowChannels' to list currently defined channels
2612 and some information about them.
2613 * Added a new action 'SIPshowregistry' to list SIP registrations.
2614 * Added TLS support for the manager interface and HTTP server
2615 * Added the URI redirect option for the built-in HTTP server
2616 * The output of CallerID in Manager events is now more consistent.
2617 CallerIDNum is used for number and CallerIDName for name.
2618 * Enable https support for builtin web server.
2619 See configs/http.conf.sample for details.
2620 * Added a new action, GetConfigJSON, which can return the contents of an
2621 Asterisk configuration file in JSON format. This is intended to help
2622 improve the performance of AJAX applications using the manager interface
2624 * SIP and IAX manager events now use "ChannelType" in all cases where we
2625 indicate channel driver. Previously, we used a mixture of "Channel"
2626 and "ChannelDriver" headers.
2627 * Added a "Bridge" action which allows you to bridge any two channels that
2628 are currently active on the system.
2629 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
2630 the voicemail users setup.
2631 * Added 'DBDel' and 'DBDelTree' manager commands.
2632 * cdr_manager now reports events via the "cdr" level, separating it from
2633 the very verbose "call" level.
2634 * Manager users are now stored in memory. If you change the manager account
2635 list (delete or add accounts) you need to reload manager.
2636 * Added Masquerade manager event for when a masquerade happens between
2638 * Added "manager reload" command for the CLI
2639 * Lots of commands that only provided information are now allowed under the
2640 Reporting privilege, instead of only under Call or System.
2641 * The IAX* commands now require either System or Reporting privilege, to
2642 mirror the privileges of the SIP* commands.
2643 * Added ability to retrieve list of categories in a config file.
2644 * Added ability to retrieve the content of a particular category.
2645 * Added ability to empty a context.
2646 * Created new action to create a new file.
2647 * Updated delete action to allow deletion by line number with respect to category.
2648 * Added new action insert to add new variable to category at specified line.
2649 * Updated action newcat to allow new category to be inserted in file above another
2651 * Added new event "JitterBufStats" in the IAX2 channel
2652 * Originate now requires the Originate privilege and, if you want to call out
2653 to a subshell, it requires the System privilege, as well. This was done to
2654 enhance manager security.
2655 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
2656 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
2657 or manager show command Atxfer from the CLI
2658 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
2659 details or manager show command IAXregistry from the CLI
2663 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
2664 state in the dialplan, as well as creating custom device states that are
2665 controllable from the dialplan.
2666 * Extend CALLERID() function with "pres" and "ton" parameters to
2667 fetch string representation of calling number presentation indicator
2668 and numeric representation of type of calling number value.
2669 * MailboxExists converted to dialplan function
2670 * A new option to Dial() for telling IP phones not to count the call
2671 as "missed" when dial times out and cancels.
2672 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
2673 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
2674 held for any given channel. Also, locks are automatically freed when a
2676 * Added HINT() dialplan function that allows retrieving hint information.
2677 Hints are mappings between extensions and devices for the sake of
2678 determining the state of an extension. This function can retrieve the list
2679 of devices or the name associated with a hint.
2680 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
2682 * Added SYSINFO() dialplan function which allows retrieval of system information
2683 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
2684 the existence of a dialplan target.
2685 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
2686 upper and lower case, respectively.
2687 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
2688 ID for the call (not the Asterisk call ID or unique ID), provided that the
2689 channel driver supports this. For SIP, you get the SIP call-ID for the
2690 bridged channel which you can store in the CDR with a custom field.
2694 * Added CLI permissions, config file: cli_permissions.conf
2695 default is to allow all commands for every local user/group.
2696 Also this new feature added three new CLI commands:
2697 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
2698 - cli reload permissions
2699 - cli show permissions
2700 * New CLI command "core show hint" (usage: core show hint <exten>)
2701 * New CLI command "core show settings"
2702 * Added 'core show channels count' CLI command.
2703 * Added the ability to set the core debug and verbose values on a per-file basis.
2704 * Added 'queue pause member' and 'queue unpause member' CLI commands
2705 * Ability to set process limits ("ulimit") without restarting Asterisk
2706 * Enhanced "agi debug" to print the channel name as a prefix to the debug
2707 output to make debugging on busy systems much easier.
2708 * New CLI commands "dialplan set extenpatternmatching true/false"
2709 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
2710 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
2711 listed in the startup_commands section of cli.conf will get executed.
2712 * Added a CLI command, "devstate change", which allows you to set custom device
2713 states from the func_devstate module that provides the DEVICE_STATE() function
2714 and handling of the "Custom:" devices.
2715 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
2716 sorted into the different possible callbacks, with the number of entries
2717 currently scheduled for each. Gives you a feel for how busy the sip channel
2719 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
2720 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
2721 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
2725 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
2726 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
2727 for a received call. If it is detected, the channel will jump to the
2728 'fax' extension in the dialplan.
2729 * The default SIP useragent= identifier now includes the Asterisk version
2730 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
2731 If set, and the incoming request carries authentication info,
2732 the username to match in the users list is taken from the Digest header
2733 rather than from the From: field. This feature is considered experimental.
2734 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
2735 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
2736 * The "localmask" setting was removed in version 1.2 and the reminder about it
2737 being removed is now also removed.
2738 * A new option "busylevel" for setting a level of calls where asterisk reports
2739 a device as busy, to separate it from call-limit. This value is also added
2740 to the SIP_PEER dialplan function.
2741 * A new realtime family called "sipregs" is now supported to store SIP registration
2742 data. If this family is defined, "sippeers" will be used for configuration and
2743 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
2744 registration data, as before.
2745 * The SIPPEER function have new options for port address, call and pickup groups
2746 * Added support for T.140 realtime text in SIP/RTP
2747 * The "checkmwi" option has been removed from sip.conf, as it is no longer
2748 required due to the restructuring of how MWI is handled. See the descriptions
2749 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
2750 for more information.
2751 * Added rtpdest option to CHANNEL() dialplan function.
2752 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
2753 * SIP now adds a header to the CANCEL if the call was answered by another phone
2754 in the same dial command, or if the new c option in dial() is used.
2755 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
2756 states it is not needed. For phones, however, that do require it the "registertrying" option
2757 has been added so it can be enabled.
2758 * A new option called "callcounter" (global/peer/user level) enables call counters needed
2759 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
2760 used to enable this functionality).
2761 * New settings for timer T1 and timer B on a global level or per device. This makes it
2762 possible to force timeout faster on non-responsive SIP servers. These settings are
2763 considered advanced, so don't use them unless you have a problem.
2764 * Added a dial string option to be able to set the To: header in an INVITE to any
2766 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
2767 the qualify frequency.
2768 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
2769 were not properly torn down due to network or endpoint failures during an established
2771 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
2772 and configs/sip.conf.sample for more information on how it is used.
2773 * Added a new configuration option "authfailureevents" that enables manager events when
2774 a peer can't authenticate properly.
2775 * Added DNS manager support to registrations for peers not referencing a peer entry.
2779 * Added the trunkmaxsize configuration option to chan_iax2.
2780 * Added the srvlookup option to iax.conf
2781 * Added support for OSP. The token is set and retrieved through the CHANNEL()
2784 XMPP Google Talk/Jingle changes
2785 -------------------------------
2786 * Added the bindaddr option to gtalk.conf.
2790 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
2791 * Proper codec support in chan_skinny.
2792 * Added settings for IP and Ethernet QoS requests
2796 * Added separate settings for media QoS in mgcp.conf
2798 Console Channel Driver changes
2799 ------------------------------
2800 * Added experimental support for video send & receive to chan_oss.
2801 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
2804 Phone channel changes (chan_phone)
2805 ----------------------------------
2806 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
2808 H.323 channel Changes
2809 ---------------------
2810 * H323 remote hold notification support added (by NOTIFY message
2811 and/or H.450 supplementary service)
2813 Local channel changes
2814 ---------------------
2815 * The device state functionality in the Local channel driver has been updated
2816 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
2817 to just UNKNOWN if the extension exists.
2818 * Added jitterbuffer support for chan_local. This allows you to use the
2819 generic jitterbuffer on incoming calls going to Asterisk applications.
2820 For example, this would allow you to use a jitterbuffer for an incoming
2821 SIP call to Voicemail by putting a Local channel in the middle. This
2822 feature is enabled by using the 'j' option in the Dial string to the Local
2823 channel in conjunction with the existing 'n' option for local channels.
2824 * A 'b' option has been added which causes chan_local to return the actual channel
2825 that is behind it when queried. This is useful for transfer scenarios as the
2826 actual channel will be transferred, not the Local channel.
2828 Agent channel changes
2829 ----------------------
2830 * The ackcall and endcall options are now supplemented with options acceptdtmf
2831 and enddtmf. These allow for the DTMF keypress to be configurable. The options
2832 default to their old hard-coded values ('#' and '*' respectively) so this should
2833 not break any existing agent installations.
2835 DAHDI channel driver (chan_dahdi) Changes
2836 ----------------------------------------
2837 * SS7 support (via libss7 library)
2838 * In India, some carriers transmit CID via dtmf. Some code has been added
2839 that will handle some situations. The cidstart=polarity_IN choice has been added for
2840 those carriers that transmit CID via dtmf after a polarity change.
2841 * CID matching information is now shown when doing 'dialplan show'.
2842 * Added dahdi show version CLI command.
2843 * Added setvar support to chan_dahdi.conf channel entries.
2844 * Added two new options: mwimonitor and mwimonitornotify. These options allow
2845 you to enable MWI monitoring on FXO lines. When the MWI state changes,
2846 the script specified in the mwimonitornotify option is executed. An internal
2847 event indicating the new state of the mailbox is also generated, so that
2848 the normal MWI facilities in Asterisk work as usual.
2849 * Added signalling type 'auto', which attempts to use the same signalling type
2850 for a channel as configured in DAHDI. This is primarily designed for analog
2851 ports, but will also work for digital ports that are configured for FXS or FXO
2852 signalling types. This mode is also the default now, so if your chan_dahdi.conf
2853 does not specify signalling for a channel (which is unlikely as the sample
2854 configuration file has always recommended specifying it for every channel) then
2855 the 'auto' mode will be used for that channel if possible.
2856 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
2857 state for a channel; also ensured that the DNDState Manager event is
2858 emitted no matter how the DND state is set or cleared.
2862 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
2863 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
2864 for details. This new channel driver allows you to use Nortel i2002,
2865 i2004, and i2050 phones with Asterisk.
2866 * Added a new channel driver, chan_console, which uses portaudio as a cross
2867 platform audio interface. It was written as a channel driver that would
2868 work with Mac CoreAudio, but portaudio supports a number of other audio
2869 interfaces, as well. Note that this channel driver requires v19 or higher
2870 of portaudio; older versions have a different API.
2874 * Added the ability to specify arguments to the Dial application when using
2875 the DUNDi switch in the dialplan.
2876 * Added the ability to set weights for responses dynamically. This can be
2877 done using a global variable or a dialplan function. Using the SHELL()
2878 function would allow you to have an external script set the weight for
2880 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
2881 functions will allow you to initiate a DUNDi query from the dialplan,
2882 find out how many results there are, and access each one.
2883 * Added the ability to specifiy a port for a dundi peer.
2887 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
2888 functions will allow you to initiate an ENUM lookup from the dialplan,
2889 and Asterisk will cache the results. ENUMRESULT can be used to access
2890 the results without doing multiple DNS queries.
2894 * Added the ability to customize which sound files are used for some of the
2895 prompts within the Voicemail application by changing them in voicemail.conf
2896 * Added the ability for the "voicemail show users" CLI command to show users
2897 configured by the dynamic realtime configuration method.
2898 * MWI (Message Waiting Indication) handling has been significantly
2899 restructured internally to Asterisk. It is now totally event based
2900 instead of polling based. The voicemail application will notify other
2901 modules that have subscribed to MWI events when something in the mailbox
2903 This also means that if any other entity outside of Asterisk is changing
2904 the contents of mailboxes, then the voicemail application still needs to
2905 poll for changes. Examples of situations that would require this option
2906 are web interfaces to voicemail or an email client in the case of using
2907 IMAP storage. So, two new options have been added to voicemail.conf
2908 to account for this: "pollmailboxes" and "pollfreq". See the sample
2909 configuration file for details.
2910 * Added "tw" language support
2911 * Added support for storage of greetings using an IMAP server
2912 * Added ability to customize forward, reverse, stop, and pause keys for message playback
2913 * SMDI is now enabled in voicemail using the smdienable option.
2914 * A "lockmode" option has been added to asterisk.conf to configure the file
2915 locking method used for voicemail, and potentially other things in the
2916 future. The default is the old behavior, lockfile. However, there is a
2917 new method, "flock", that uses a different method for situations where the
2918 lockfile will not work, such as on SMB/CIFS mounts.
2919 * Added the ability to backup deleted messages, to ease recovery in the case
2920 that a user accidentally deletes a message, and discovers that they need it.
2921 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
2922 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
2923 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
2924 voicemail boxes. The SMDI interface can also poll for MWI changes when some
2925 outside entity is modifying the state of the mailbox (such as IMAP storage or
2926 a web interface of some kind).
2927 * Added the support for marking messages as "urgent." There are two methods to accomplish
2928 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
2929 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
2930 the message as urgent after he has recorded a voicemail by following the voice instructions.
2931 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
2936 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
2937 used across multiple queues.
2938 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
2939 setqueueentryvar options for each queue, see queues.conf.sample for details.
2940 * Added keepstats option to queues.conf which will keep queue
2941 statistics during a reload.
2942 * setinterfacevar option in queues.conf also now sets a variable
2943 called MEMBERNAME which contains the member's name.
2944 * Added 'Strategy' field to manager event QueueParams which represents
2945 the queue strategy in use.
2946 * Added option to run macro when a queue member is connected to a caller,
2947 see queues.conf.sample for details.
2948 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
2949 does not count paused queue members as unavailable.
2950 * Added min-announce-frequency option to queues.conf which allows you to control the
2951 minimum amount of time between queue announcements for use when the caller's queue
2952 position changes frequently.
2953 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
2955 * Added ability for non-realtime queues to have realtime members
2956 * Added the "linear" strategy to queues.
2957 * Added the "wrandom" strategy to queues.
2958 * Added new channel variable QUEUE_MIN_PENALTY
2959 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
2960 rules in queuerules.conf. See configs/queuerules.conf.sample for details
2961 * Added a new parameter for member definition, called state_interface. This may be
2962 used so that a member may be called via one interface but have a different interface's
2963 device state reported.
2964 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
2965 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
2966 "manager show command QueueReset."
2967 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
2968 specified by the periodic-announce option, then one will be chosen randomly when it is time
2969 to play a periodic announcment
2970 * New configuration options: announce-position now takes two more values in addition to "yes" and
2971 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
2972 announce-position-limit. By setting announce-position to "limit" callers will only have their
2973 position announced if their position is less than what is specified by announce-position-limit.
2974 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
2975 will be told that their are more than announce-position-limit callers waiting.
2976 * Two new queue log events have been added. An ADDMEMBER event will be logged
2977 when a realtime queue member is added and a REMOVEMEMBER event will be logged
2978 when a realtime queue member is removed. Since there is no calling channel associated
2979 with these events, the string "REALTIME" is placed where the channel's unique id
2980 is typically placed.
2981 * The configuration method for the "joinempty" and "leavewhenempty" options has
2982 changed to a comma-separated list of methods of determining member availability
2983 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
2984 values are still accepted for backwards-compatibility, though.
2985 * The average talktime is now calculated on queues. This information is reported via the
2986 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
2987 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
2992 * The 'o' option to provide an optimization has been removed and its functionality
2993 has been enabled by default.
2994 * When a conference is created, the UNIQUEID of the channel that caused it to be
2995 created is stored. Then, every channel that joins the conference will have the
2996 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
2997 callers that come and go from long standing conferences.
2998 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
2999 except it does operations on a channel by name, instead of number in a conference.
3000 This is a very useful feature in combination with the 'X' option to ChanSpy.
3001 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
3003 * Added new RealTime functionality to provide support for scheduled conferencing.
3004 This includes optional messages to the caller if they attempt to join before
3005 the schedule start time, or to allow the caller to join the conference early.
3006 Also included is optional support for limiting the number of callers per
3007 RealTime conference.
3008 * Added the S() and L() options to the MeetMe application. These are pretty
3009 much identical to the S() and L() options to Dial(). They let you set
3010 timeouts for the conference, as well as have warning sounds played to
3011 let the caller know how much time is left, and when it is running out.
3012 * Added the ability to do "meetme concise" with the "meetme" CLI command.
3013 This extends the concise capabilities of this CLI command to include
3014 listing all conferences, instead of an addition to the other sub commands
3015 for the "meetme" command.
3016 * Added the ability to specify the music on hold class used to play into the
3017 conference when there is only one member and the M option is used.
3018 * Added MEETME_INFO dialplan function which provides a way to query
3019 various properties of a Meetme conference.
3020 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
3021 and *84: record in-conf
3023 Other Dialplan Application Changes
3024 ----------------------------------
3025 * Argument support for Gosub application
3026 * From the to-do lists: straighten out the app timeout args:
3027 Wait() app now really does 0.3 seconds- was truncating arg to an int.
3028 WaitExten() same as Wait().
3029 Congestion() - Now takes floating pt. argument.
3030 Busy() - now takes floating pt. argument.
3031 Read() - timeout now can be floating pt.
3032 WaitForRing() now takes floating pt timeout arg.
3033 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
3034 * Added 's' option to Page application.
3035 * Added an optional timeout argument to the Page application.
3036 * Added 'E', 'V', and 'P' commands to ExternalIVR.
3037 * Added 'o' and 'X' options to Chanspy.
3038 * Added a new dialplan application, Bridge, which allows you to bridge the
3039 calling channel to any other active channel on the system.
3040 * Added the ability to specify a music on hold class to play instead of ringing
3041 for the SLATrunk application.
3042 * The Read application no longer exits the dialplan on error. Instead, it sets
3043 READSTATUS to ERROR, which you can catch and handle separately.
3044 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
3045 of asking for verification of each name, one at a time.
3046 * Privacy() no longer uses privacy.conf, as all options are specifyable as
3047 direct options to the app.
3048 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
3050 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
3051 * The ChannelRedirect application no longer exits the dialplan if the given channel
3052 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
3053 or NOCHANNEL if the given channel was not found.
3054 * The silencethreshold setting that was previously configurable in multiple
3055 applications is now settable globally via dsp.conf.
3057 Music On Hold Changes
3058 ---------------------
3059 * A new option, "digit", has been added for music on hold classes in
3060 musiconhold.conf. If this is set for a music on hold class, a caller
3061 listening to music on hold can press this digit to switch to listening
3062 to this music on hold class.
3063 * Support for realtime music on hold has been added.
3064 * In conjunction with the realtime music on hold, a general section has
3065 been added to musiconhold.conf, its sole variable is cachertclasses. If this
3066 is set, then music on hold classes found in realtime will be cached in memory.
3070 * AEL upgraded to use the Gosub with Arguments instead
3071 of Macro application, to hopefully reduce the problems
3072 seen with the artificially low stack ceiling that
3073 Macro bumps into. Macros can only call other Macros
3074 to a depth of 7. Tests run using gosub, show depths
3075 limited only by virtual memory. A small test demonstrated
3076 recursive call depths of 100,000 without problems.
3077 -- in addition to this, all apps that allowed a macro
3078 to be called, as in Dial, queues, etc, are now allowing
3079 a gosub call in similar fashion.
3080 * AEL now generates LOCAL(argname) declarations when it
3081 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
3082 etc. That makes the arguments local in scope. The user
3083 can define their own local variables in macros, now,
3084 by saying "local myvar=someval;" or using Set() in this
3085 fashion: Set(LOCAL(myvar)=someval); ("local" is now
3087 * utils/conf2ael introduced. Will convert an extensions.conf
3088 file into extensions.ael. Very crude and unfinished, but
3089 will be improved as time goes by. Should be useful for a
3090 first pass at conversion.
3091 * aelparse will now read extensions.conf to see if a referenced
3092 macro or context is there before issueing a warning.
3093 * AEL parser sets a local channel variable ~~EXTEN~~, to
3094 preserve the value of ${EXTEN} thru switch statements.
3095 * New operator in $[...] expressions: the ~~ operator serves
3096 as a concatenation operator. AT THE MOMENT, it is really only
3097 necessary and useful in AEL, especially in if() expressions.
3098 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
3099 any enclosing double-quotes, and evaluate to the value of a
3100 concatenated with the value of b. For example if a is set to
3101 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
3102 evaluate to xyzabc .
3105 Call Features (res_features) Changes
3106 ------------------------------------
3107 * Added the parkedcalltransfers option to features.conf
3108 * Added parkedcallparking option to control one touch parking w/ parking
3110 * Added parkedcallhangup option to control disconnect feature w/ parking
3112 * Added parkedcallrecording option to control one-touch record w/ parking
3114 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
3115 parkedcalltransfers option support for multiple parking lots.
3116 * Added BRIDGE_FEATURES variable to set available features for a channel
3117 * The built-in method for doing attended transfers has been updated to
3118 include some new options that allow you to have the transferee sent
3119 back to the person that did the transfer if the transfer is not successful.
3120 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
3121 in features.conf.sample.
3122 * Added support for configuring named groups of custom call features in
3123 features.conf. This means that features can be written a single time, and
3124 then mapped into groups of features for different key mappings or easier
3126 * Updated the ParkedCall application to allow you to not specify a parking
3127 extension. If you don't specify a parking space to pick up, it will grab
3128 the first one available.
3129 * Added cli command 'features reload' to reload call features from features.conf
3130 * Moved into core asterisk binary.
3131 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
3132 * Added the ability for custom parking lots to be configured with their own
3133 parking extension with the parkext option.
3135 Language Support Changes
3136 ------------------------
3137 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
3138 * Added support for the Hungarian language for saying numbers, dates, and times.
3142 * Added SPEECH commands for speech recognition. A complete listing can be found
3144 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
3145 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
3146 does not behave as expected; the native command needs to be used, instead.
3147 * Added the ability to perform SRV lookups on fast AGI calls. To use this
3148 feature, simply use hagi: instead of agi: as the protocol portion
3149 of the URI parameter to the AGI function call in your dial plan. Also note
3150 that specifying a port number in the AGI URI will disable SRV lookups,
3151 even if you use the hagi: protocol.
3152 * No longer support MSG_OOB flag on HANGUP.
3156 * Added rotatestrategy option to logger.conf, along with two new options:
3157 "timestamp" which will use the time to name the logger files instead of
3158 sequence number; and "rotate", which rotates the names of the log files,
3159 similar to the way syslog rotates files.
3160 * Added exec_after_rotate option to logger.conf, which allows a system
3161 command to be run after rotation. This is primarily useful with
3162 rotatestrategy=rotate, to allow a limit on the number of log files kept
3163 and to ensure that the oldest log file gets deleted.
3164 * Added realtime support for the queue log
3168 * The cdr_manager module has a [mappings] feature, like cdr_custom,
3169 to add fields to the manager event from the CDR variables.
3170 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
3171 backend database CDR table. Specifically, additional, non-standard
3172 columns are supported, merely by setting the corresponding CDR variable in
3173 your dialplan. In addition, you may alias any column to another name (for
3174 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
3175 simply "alias src => ANI" in the configuration file). Records may be
3176 posted to more than one backend, simply by specifying multiple categories
3177 in the configuration file. And finally, you may filter which CDRs get
3178 posted to each backend, by specifying a filter (which the record must
3179 match) for the particular category. Filters are additive (meaning all
3180 rules must match to post that CDR).
3181 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
3182 module. Specifically, you may add additional columns into the table and
3183 they will be set, if you set the corresponding CDR variable name. Also,
3184 if you omit columns in your database table, they will be silently skipped
3185 (but a record will still be inserted, based on what columns remain). Note
3186 that the other two features from cdr_adaptive_odbc (alias and filter) are
3187 not currently supported.
3188 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
3189 has been disabled using the NoCDR application.
3191 Miscellaneous New Modules
3192 -------------------------
3193 * Added a new CDR module, cdr_sqlite3_custom.
3194 * Added a new realtime configuration module, res_config_sqlite
3195 * Added a new codec translation module, codec_resample, which re-samples
3196 signed linear audio between 8 kHz and 16 kHz to help support wideband
3198 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
3199 based on configuration templates that use Asterisk dialplan function and
3200 variable substitution. It should be possible to create phone profiles and
3201 templates that work for the majority of phones provisioned over http. It
3202 is currently only intended to provision a single user account per phone.
3203 An example profile and set of templates for Polycom phones is provided.
3204 NOTE: Polycom firmware is not included, but should be placed in
3205 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
3206 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
3207 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
3208 provided; there is a JACK() application, and a JACK_HOOK() function. Both
3209 interfaces create an input and output JACK port. The application makes
3210 these ports the endpoint of the call. The audio coming from the channel
3211 goes out the output port and whatever comes back in on the input port is
3212 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
3213 audiohook on the channel. This lets you run the audio coming from a
3214 channel through JACK, and whatever comes back in is what gets forwarded
3215 on as the channel's audio. This is very useful for building custom
3216 vocoders or doing recording or analysis of the channel's audio in another
3218 * Added a new module, res_config_curl, which permits using a HTTP POST url
3219 to retrieve, create, update, and delete realtime information from a remote
3220 web server. Note that this module requires func_curl.so to be loaded for
3221 backend functionality.
3222 * Added a new module, res_config_ldap, which permits the use of an LDAP
3223 server for realtime data access.
3224 * Added support for writing and running your dialplan in lua using the pbx_lua
3225 module. See configs/extensions.lua.sample for examples of how to do this.
3229 * Ability to use libcap to set high ToS bits when non-root
3230 on Linux. If configure is unable to find libcap then you
3231 can use --with-cap to specify the path.
3232 * Added maxfiles option to options section of asterisk.conf which allows you to specify
3233 what Asterisk should set as the maximum number of open files when it loads.
3234 * Added the jittertargetextra configuration option.
3235 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
3236 configuration files for the IP channel drivers. The new option is "cos".
3237 This information is also documented on the Asterisk wiki at
3238 https://wiki.asterisk.org/wiki/x/EYBG
3239 * When originating a call using AMI or pbx_spool that fails the reason for failure
3240 will now be available in the failed extension using the REASON dialplan variable.
3241 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
3242 It allows you to configure a prefix for auto-monitor recordings.
3243 * A new extension pattern matching algorithm, based on a trie, is introduced
3244 here, that could noticeably speed up mid-sized to large dialplans.
3245 It is NOT used by default, as duplicating the behaviour of the old pattern
3246 matcher is still under development. A config file option, in extensions.conf,
3247 in the [general] section, called "extenpatternmatchingnew", is by default
3248 set to false; setting that to true will force the use of the new algorithm.
3249 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
3250 be used to switch the algorithms at run time.
3251 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
3252 specifying which socket to use to connect to the running Asterisk daemon
3254 * Performance enhancements to the sched facility, which is used in
3255 the channel drivers, etc. Added hashtabs and doubly-linked lists
3256 to speed up deletion; start at the beginning or end of list to
3258 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
3259 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
3260 Added regression tests to the tests/ dir, also.
3261 * Added a refcount trace feature to astobj2 for those trying to balance
3262 object creation, deletion; work, play; space and time. See the
3263 notes in astobj2.h. Also, see utils/refcounter as well, as a
3264 quick way to find unbalanced refcounts in what could be a sea
3265 of objects that were balanced.
3266 * Added logging to 'make update' command. See update.log
3267 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
3268 do not come from the remote party.
3269 * Added the 'n' option to the SpeechBackground application to tell it to not
3270 answer the channel if it has not already been answered.
3271 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
3272 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
3274 * iLBC source code no longer included (see UPGRADE.txt for details)
3275 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
3276 deadlock is detected, a backtrace of the stack which led to the lock calls
3277 will be output to the CLI.
3278 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
3279 the "core show locks" CLI command will give lock information output as well
3280 as a backtrace of the stack which led to the lock calls.
3281 * users.conf now sports an optional alternateexts property, which permits
3282 allocation of additional extensions which will reach the specified user.
3283 * A new option for the configure script, --enable-internal-poll, has been added
3284 for use with systems which may have a buggy implementation of the poll system
3285 call. If you notice odd behavior such as the CLI being unresponsive on remote
3286 consoles, you may want to try using this option. This option is enabled by default
3287 on Darwin systems since it is known that the Darwin poll() implementation has
3291 --------------------
3292 * In addition to timing from DAHDI, there is a new timing module called
3293 res_timing_timerfd. In order to use this, you must be running Linux with
3294 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
3295 script will be able to tell if you have the requirements. From menuselect, select
3296 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob