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 'c' option has been removed. It is not possible to modify the name of a
20 channel involved in a CDR.
24 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
27 * Variables are no longer purged from the original CDR. See the 'v' option for
30 * The 'A' option has been removed. The Answer time on a CDR is never updated
33 * The 'd' option has been removed. The disposition on a CDR is a function of
34 the state of the channel and cannot be altered.
36 * The 'D' option has been removed. Who the Party B is on a CDR is a function
37 of the state of the respective channels, and cannot be altered.
39 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
40 such that the start time and, if applicable, the answer time was updated.
41 Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
42 'r' option now triggers the Reset, setting the start time (and answer time
43 if applicable) to the current time.
45 * The 's' option has been removed. A variable can be set on the original CDR
46 if desired using the CDR function, and removed from a forked CDR using the
49 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
50 longer applies in the CDR engine.
52 * The 'v' option now prevents the copy of the variables from the original CDR
53 to the forked CDR. Previously the variables were always copied but were
54 removed from the original. Removing variables from a CDR can have unintended
55 side effects - this option allows the user to prevent propagation of
56 variables from the original to the forked without modifying the original.
60 * Added the 'n' option to MeetMe to prevent application of the DENOISE function
61 to a channel joining a conference. Some channel drivers that vary the number
62 of audio samples in a voice frame will experience significant quality problems
63 if a denoiser is attached to the channel; this option gives them the ability
64 to remove the denoiser without having to unload func_speex.
68 * The NoCDR application is deprecated. Please use the CDR_PROP function to
70 * While the NoCDR application will prevent CDRs for a channel from being
71 propagated to registered CDR backends, it will not prevent that data from
72 being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
73 function that enables CDRs on a channel will restore those records that have
74 not yet been finalized.
78 * Add queue available hint. exten => 8501,hint,Queue:markq_avail
79 Note: the suffix '_avail' after the queuename.
80 Reports 'InUse' for no logged in agents or no free agents.
81 Reports 'Idle' when an agent is free.
83 * The configuration options eventwhencalled and eventmemberstatus have been
84 removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
85 AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
86 sent. The "Variable" fields will also no longer exist on the Agent* events.
90 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
91 CDRs when they were previously disabled on a channel.
92 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
93 backends occurs on an as-needed basis in order to preserve linkedid
94 propagation and other needed behavior.
98 * This application is deprecated in favor of the CHANNEL function.
102 * UserEvent will now handle duplicate keys by overwriting the previous value
103 assigned to the key. UserEvent invocations will also be distributed to any
104 interested res_stasis applications.
109 * Redirecting reasons can now be set to arbitrary strings. This means
110 that the REDIRECTING dialplan function can be used to set the redirecting
111 reason to any string. It also allows for custom strings to be read as the
112 redirecting reason from SIP Diversion headers.
114 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
115 must be on the channel initiating the transfer to have any effect.
117 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
118 driver specific. If the channel variable is set on the transferrer channel,
119 the sound will be played to the target of an attended transfer.
121 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
122 a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
123 listed. Any more peers in the bridge will not be included in the list.
124 BRIDGEPEER is not valid in holding bridges like parking since those channels
125 do not talk to each other even though they are in a bridge.
127 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
128 and will contain a value if the BRIDGEPEER's channel driver supports it.
130 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
131 removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
132 activated the dynamic feature.
134 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
135 only on the channel executing the dynamic feature. Executing a dynamic
136 feature on the bridge peer in a multi-party bridge will execute it on all
137 peers of the activating channel.
139 AMI (Asterisk Manager Interface)
141 * The SIPshowpeer action will now include a 'SubscribeContext' field for a peer
142 in its response if the peer has a subscribe context set.
144 * The SIPqualifypeer action now acknowledges the request once it has established
145 that the request is against a known peer. It also issues a new event,
146 'SIPqualifypeerdone', once the qualify action has been completed.
148 * The PlayDTMF action now supports an optional 'Duration' parameter. This
149 specifies the duration of the digit to be played, in milliseconds.
151 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
152 updates when changes occur instead of requiring the use of pollmailboxes.
154 * CLI Command 'Manager Show Commands' no longer truncates command names longer
155 than 15 characters and no longer shows authorization requirement for commands.
156 'Manager Show Command' now displays the privileges needed for using a given
157 manager command instead.
159 * Added new action "ControlPlayback". The ControlPlayback action allows an AMI
160 client to manipulate audio currently being played back on a channel. The
161 supported operations depend on the application being used to send audio to
162 the channel. When the audio playback was initiated using the ControlPlayback
163 application or CONTROL STREAM FILE AGI command, the audio can be paused,
164 stopped, restarted, reversed, or skipped forward. When initiated by other
165 mechanisms (such as the Playback application), the audio can be stopped,
166 reversed, or skipped forward.
168 * Channel related events now contain a snapshot of channel state, adding new
169 fields to many of these events.
171 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
172 in a future release. Please use the common 'Exten' field instead.
174 * The AMI event 'UserEvent' from app_userevent now contains the channel state
175 fields. The channel state fields will come before the body fields.
177 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
178 'UnParkedCall' have changed significantly in the new res_parking module.
179 First, channel snapshot data is included for both the parker and the parkee
180 in lieu of the "From" and "Channel" fields. They follow standard channel
181 snapshot format but each field is suffixed with 'Parker' or 'Parkee'
182 depending on which side it applies to. The 'Exten' field is replaced with
183 'ParkingSpace' since the registration of extensions as for parking spaces
184 is no longer mandatory.
186 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
187 fashion has changed the field names 'StartExten' and 'StopExten' to
188 'StartSpace' and 'StopSpace' respectively.
190 * The deprecated use of | (pipe) as a separator in the channelvars setting in
191 manager.conf has been removed.
193 * Channel Variables conveyed with a channel no longer contain the name of the
194 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
195 ChanVariable: bar=baz. When multiple channels are present in a single AMI
196 event, the various ChanVariable fields will contain a suffix that specifies
197 which channel they correspond to.
199 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
200 event always conveys the AMI event for a particular channel.
202 * All "Reload" events have been consolidated into a single event type. This
203 event will always contain a Module field specifying the name of the module
204 and a Status field denoting the result of the reload. All modules now issue
205 this event when being reloaded.
207 * The "ModuleLoadReport" event has been removed. Most AMI connections would
208 fail to receive this event due to being connected after modules have loaded.
209 AMI connections that want to know when Asterisk is ready should listen for
210 the "FullyBooted" event.
212 * app_fax now sends the same send fax/receive fax events as res_fax. The
213 "FaxSent" event is now the "SendFAX" event, and the "FaxReceived" event is
214 now the "ReceiveFAX" event.
216 * The MusicOnHold event is now two events: MusicOnHoldStart and
217 MusicOnHoldStop. The sub type field has been removed.
219 * The JabberEvent event has been removed. It is not AMI's purpose to be a
220 carrier for another protocol.
222 * The Bridge Manager action's Playtone header now accepts more fine-grained
223 options. "Channel1" and "Channel2" may be specified in order to play a tone
224 to the specific channel. "Both" may be specified to play a tone to both
225 channels. The old "yes" option is still accepted as a way of playing the
226 tone to Channel2 only.
228 * The AMI 'Status' response event to the AMI Status action replaces the
229 BridgedChannel and BridgedUniqueid headers with the BridgeID header to
230 indicate what bridge the channel is currently in.
232 * The AMI 'Hold' event has been moved out of individual channel drivers, into
233 core, and is now two events: Hold and Unhold. The status field has been
236 * The AMI events in app_queue have been made more consistent with each other.
237 Events that reference channels (QueueCaller* and Agent*) will show
238 information about each channel. The (infamous) "Join" and "Leave" AMI
239 events have been changed to "QueueCallerJoin" and "QueueCallerLeave".
241 AGI (Asterisk Gateway Interface)
243 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
245 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
248 CDR (Call Detail Records)
250 * Significant changes have been made to the behavior of CDRs. For a full
251 definition of CDR behavior in Asterisk 12, please read the specification
252 on the Asterisk wiki (wiki.asterisk.org).
254 * CDRs will now be created between all participants in a bridge. For each
255 pair of channels in a bridge, a CDR is created to represent the path of
256 communication between those two endpoints. This lets an end user choose who
257 to bill for what during multi-party bridges or bridge operations during
260 * When a CDR is dispatched, user defined CDR variables from both parties are
261 included in the resulting CDR. If both parties have the same variable, only
262 the Party A value is provided.
266 * The BRIDGE_FEATURES channel variable would previously only set features for
267 the calling party and would set this feature regardless of whether the
268 feature was in caps or in lowercase. Use of a caps feature for a letter
269 will now apply the feature to the calling party while use of a lowercase
270 letter will apply that feature to the called party.
272 * Add support for automixmonitor to the BRIDGE_FEATURES channel variable.
274 * Parking has been pulled from core and placed into a separate module called
275 res_parking. See Parking changes below for more details.
277 * You can now have the settings for a channel updated using the FEATURE()
278 and FEATUREMAP() functions inherited to child channels by setting
279 FEATURE(inherit)=yes.
283 * When performing queue pause/unpause on an interface without specifying an
284 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
285 least one member of any queue exists for that interface.
287 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
288 for realtime queue log entries.
292 * Parking is now implemented as a module instead of as core functionality.
293 The preferred way to configure parking is now through res_parking.conf while
294 configuration through features.conf is not currently supported.
296 * res_parking uses the configuration framework. If an invalid configuration is
297 supplied, res_parking will fail to load or fail to reload. Previously parking
298 lots that were misconfigured would generally be accepted with certain
299 configuration problems leading to individual disabled parking lots.
301 * Parked calls are now placed in bridges. This is a largely architectural change,
302 but it could have some implications in allowing for new parked call retrieval
303 methods and the contents of parking lots will be visible though certain bridge
306 * The order of arguments for the new parking applications are different from the
307 old ones to be more intuitive. Timeout and return context/exten/priority are now
308 implemented as options. parking_lot_name is now the first parameter. See the
309 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
310 in-depth information as well as syntax.
312 * Extensions are no longer automatically created in the dialplan to park calls,
313 pickup parked calls, etc by default.
315 * adsipark is no longer supported under the new parking model
317 * The PARKINGSLOT channel variable has been deprecated in favor of PARKING_SPACE
318 to match the naming scheme of the new system.
320 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
321 channel even when comebactoorigin=yes
323 * New CLI command 'parking show' allows you to inspect the currently in use
324 parking lots. 'parking show <parkinglot>' will also show the parked calls
325 in that specific parking lot.
327 * The CLI command 'parkedcalls' is now deprecated in favor of
328 'parking show <parkinglot>'.
330 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
331 can be used to get a list of parked calls only for a specific parking lot.
333 * The AMI command 'Park' has had the argument 'Channel2' renamed to
334 'TimeoutChannel'. 'TimeoutChannel' is no longer a required argument.
335 'Channel2' can still be used as the argument name, but it is deprecated
336 and the 'TimeoutChannel' argument will be used if both are present.
338 * The ParkAndAnnounce application is now provided through res_parking instead
339 of through the separate app_parkandannounce module.
341 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
342 by default. Instead, it will follow the timeout rules of the parking lot. The
343 old behavior can be reproduced by using the 'c' option.
345 * Dynamic parking lots will now fail to be created if the parking lot specified
346 by PARKINGDYNAMIC does not exist.
348 * Dynamic parking lots will also fail to be created now if they require exclusive
349 park and parkedcall extensions which overlap with other parking lots.
351 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
352 currently contain no calls. Dynamic parking lots containing parked calls will
353 persist through the reloads without alteration.
355 * If parkext_exclusive is set for a parking lot and that extension is already in
356 use when that parking lot tries to register it, this is now considered a parking
357 system configuration error. Configurations which do this will be rejected.
358 Dynamic parking lots which try to register extensions that already exist will
361 * Added a channel variable PARKER_FLAT which stores the name of the extension
362 that would be used to come back to if comebacktoorigin was set to use. This can
363 be useful when comebacktoorigin is off if you still want to use the extensions
364 in the park-dial context that are generated to redial the parker on timeout.
368 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
369 will store the path information for that peer when it registers. Realtime
370 tables can also use the 'supportpath' field to enable Path header support.
372 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
373 objectIdentifier. This maps to the supportpath option in sip.conf.
377 * All future modules which utilize Sorcery for object persistence must have a
378 column named "id" within their schema when using the Sorcery realtime module.
379 This column must be able to contain a string of up to 128 characters in length.
381 Security Events Framework
382 -------------------------
383 * Security Event timestamps now use ISO 8601 formatted date/time instead of the
384 "seconds-microseconds" format that it was using previously.
389 * When a channel driver is configured to enable jiterbuffers, they are now
390 applied unconditionally when a channel joins a bridge. If a jitterbuffer
391 is already set for that channel when it enters, such as by the JITTERBUFFER
392 function, then the existing jitterbuffer will be used and the one set by
393 the channel driver will not be applied.
397 * The updatecdr option has been removed. Altering the names of channels on a
398 CDR is not supported - the name of the channel is the name of the channel,
399 and pretending otherwise helps no one.
400 * The AGENTUPDATECDR channel variable has also been removed, for the same
401 reason as the updatecdr option.
405 * The /b option is removed.
407 * chan_local moved into the system core and is no longer a loadable module.
411 * Added general support for busy detection.
413 * Added ECAM command support for Sony Ericsson phones.
417 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
418 using the 'supportpath' setting, either on a global basis or on a peer basis.
419 This setting enables Asterisk to route outgoing out-of-dialog requests via a
420 set of proxies by using a pre-loaded route-set defined by the Path headers in
421 the REGISTER request. See Realtime updates for more configuration information.
429 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
430 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
431 The value of this setting is ignored when disabled is used for the argument.
435 * The 'amaflags' and 'accountcode' attributes for the CDR function are
436 deprecated. Use the CHANNEL function instead to access these attributes.
437 * The 'l' option has been removed. When reading a CDR attribute, the most
438 recent record is always used. When writing a CDR attribute, all non-finalized
440 * The 'r' option has been removed, for the same reason as the 'l' option.
441 * The 's' option has been removed, as LOCKED semantics no longer exist in the
446 * A new function CDR_PROP has been added. This function lets you set properties
447 on a channel's active CDRs. This function is write-only. Properties accept
448 boolean values to set/clear them on the channel's CDRs. Valid properties
450 * 'party_a' - make this channel the preferred Party A in any CDR between two
451 channels. If two channels have this property set, the creation time of the
452 channel is used to determine who is Party A. Note that dialed channels are
453 never Party A in a CDR.
454 * 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
455 application when set to True, and analogous to the 'e' option in ResetCDR
464 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
465 them, an Asterisk-specific version of pjproject needs to be installed.
466 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
470 * Device state for XMPP buddies is now available using the following format:
471 XMPP/<client name>/<buddy address>
472 If any resource is available the device state is considered to be not in use.
473 If no resources exist or all are unavailable the device state is considered
477 ------------------------------------------------------------------------------
478 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
479 ------------------------------------------------------------------------------
485 * The Asterisk build system will now build and install a shared library
486 (libasteriskssl.so) used to wrap various initialization and shutdown functions
487 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
488 that Asterisk can ensure that these functions do *not* get called by any
489 modules that are loaded into Asterisk, since they should only be called once
490 in any single process. If desired, this feature can be disabled by supplying
491 the "--disable-asteriskssl" option to the configure script.
493 * A new make target, 'full', has been added to the Makefile. This performs
494 the same compilation actions as make all, but will also scan the entirety of
495 each source file for documentation. This option is needed to generate AMI
496 event documentation. Note that your system must have Python in order for
497 this make target to succeed.
499 * The optimization portion of the build system has been reworked to avoid
500 broken builds on certain architectures. All architecture-specific
501 optimization has been removed in favor of using -march=native to allow gcc
502 to detect the environment in which it is running when possible. This can
503 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
505 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
506 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
508 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
509 previously parsed the header file to obtain the version of Asterisk, you
510 will now have to go through Asterisk to get the version information.
518 * Added 'F()' option. Similar to the dial option, this can be supplied with
519 arguments indicating where the callee should go after the caller is hung up,
520 or without options specified, the priority after the Queue will be used.
525 * Added menu action admin_toggle_mute_participants. This will mute / unmute
526 all non-admin participants on a conference. The confbridge configuration
527 file also allows for the default sounds played to all conference users when
528 this occurs to be overriden using sound_participants_unmuted and
529 sound_participants_muted.
531 * Added menu action participant_count. This will playback the number of
532 current participants in a conference.
534 * Added announcement configuration option to user profile. If set the sound
535 file will be played to the user, and only the user, upon joining the
538 * Added record_file_append option that defaults to "yes", but if set to no
539 will create a new file between each start/stop recording.
544 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
545 channels respectively before the callee channels are called.
550 * Added support for IPv6.
552 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
553 external process will cause the current playlist to be cleared, including
554 stopping any audio file that is currently playing. This is useful when you
555 want to interrupt audio playback only when specific DTMF is entered by the
561 * A new option, 'I' has been added to app_followme. By setting this option,
562 Asterisk will not update the caller with connected line changes when they
563 occur. This is similar to app_dial and app_queue.
565 * The 'N' option is now ignored if the call is already answered.
567 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
568 and caller channels respectively before the callee channels are called.
570 * The winning FollowMe outgoing call is now put on hold if the caller put it on
576 * MixMonitor hooks now have IDs associated with them which can be used to
577 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
578 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
579 now accepts that ID as an argument.
581 * Added 'm' option, which stores a copy of the recording as a voicemail in the
587 * The connect action in app_mysql now allows you to specify a port number to
588 connect to. This is useful if you run a MySQL server on a non-standard
594 * Increased the default number of allowed destinations from 5 to 12.
599 * The app_page application now no longer depends on DAHDI or app_meetme. It
600 has been re-architected to use app_confbridge internally.
605 * Added queue options autopausebusy and autopauseunavail for automatically
606 pausing a queue member when their device reports busy or congestion.
608 * The 'ignorebusy' option for queue members has been deprecated in favor of
609 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
610 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
611 per interface basis. Individual ringinuse values can now be set in
612 queues.conf via an argument to member definitions. Lastly, the queue
613 'ringinuse' setting now only determines defaults for the per member
614 'ringinuse' setting and does not override per member settings like it does
617 * Added 'F()' option. Similar to the dial option, this can be supplied with
618 arguments indicating where the callee should go after the caller is hung up,
619 or without options specified, the priority after the Queue will be used.
621 * Added new option log_member_name_as_agent, which will cause the membername to
622 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
623 state_interface has been set.
625 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
627 * App_queue will now play periodic announcements for the caller that
628 holds the first position in the queue while waiting for answer.
632 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
633 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
634 changed arguments to SayUnixTime so that every option is truly optional even
635 when using multiple options (so that j option could be used without having to
636 manually specify timezone and format) There are other benefits, e.g., format
637 can now be used without specifying time zone as well.
642 * Addition of the VM_INFO function - see Function changes.
644 * The imapserver, imapport, and imapflags configuration options can now be
645 overriden on a user by user basis.
647 * When voicemail plays a message's envelope with saycid set to yes, when
648 reaching the caller id field it will play a recording of a file with the same
649 base name as the sender's callerid if there is a similarly named file in
650 <astspooldir>/recordings/callerids/
652 * Voicemails now contains a unique message identifier "msg_id", which is stored
653 in the message envelope with the sound files. IMAP backends will now store
654 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
655 backends will store the message identifier in a "msg_id" column. See
656 UPGRADE.txt for more information.
658 * Added VoiceMailPlayMsg application. This application will play a single
659 voicemail message from a mailbox. The result of the application, SUCCESS or
660 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
665 * Hangup handlers can be attached to channels using the CHANNEL() function.
666 Hangup handlers will run when the channel is hung up similar to the h
667 extension. The hangup_handler_push option will push a GoSub compatible
668 location in the dialplan onto the channel's hangup handler stack. The
669 hangup_handler_pop option will remove the last added location, and optionally
670 replace it with a new GoSub compatible location. The hangup_handler_wipe
671 option will remove all locations on the stack, and optionally add a new
674 * The expression parser now recognizes the ABS() absolute value function,
675 which will convert negative floating point values to positive values.
677 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
678 control of faxdetect.
680 * Addition of the VM_INFO function that can be used to retrieve voicemail
681 user information, such as the email address and full name.
682 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
685 * The REDIRECTING function now supports the redirecting original party id
688 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
689 lets you set some of the configuration options from the [general] section
690 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
691 the key sequence used to activate built-in features, such as blindxfer,
692 and automon. See the built-in documentation for details.
694 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
695 instead of simply the uri. This is the format that MessageSend() can use
696 in the from parameter for outgoing SIP messages.
698 * Added the PRESENCE_STATE function. This allows retrieving presence state
699 information from any presence state provider. It also allows setting
700 presence state information from a CustomPresence presence state provider.
701 See AMI/CLI changes for related commands.
703 * Added the AMI_CLIENT function to make manager account attributes available
704 to the dialplan. It currently supports returning the current number of
705 active sessions for a given account.
707 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
708 and the REDIRECTING functions.
716 * Added a manager event "LocalBridge" for local channel call bridges between
717 the two pseudo-channels created.
722 * Added dialtone_detect option for analog ports to disconnect incoming
723 calls when dialtone is detected.
725 * Added option colp_send to send ISDN connected line information. Allowed
726 settings are block, to not send any connected line information; connect, to
727 send connected line information on initial connect; and update, to send
728 information on any update during a call. Default is update.
730 * Add options namedcallgroup and namedpickupgroup to support installations
731 where a higher number of groups (>64) is required.
733 * Added support to use private party ID information with PRI calls.
738 * A new channel driver named chan_motif has been added which provides support for
739 Google Talk and Jingle in a single channel driver. This new channel driver includes
740 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
741 hold, unhold, and ringing notification. It is also compliant with the current Jingle
742 specification, current Google Jingle specification, and the original Google Talk
748 * Added NAT support for RTP. Setting in config is 'nat', which can be set
749 globally and overriden on a peer by peer basis.
751 * Direct media functionality has been added. Options in config are:
752 directmedia (directrtp) and directrtpsetup (earlydirect)
754 * ChannelUpdate events now contain a CallRef header.
759 * Asterisk will no longer substitute CID number for CID name in the display
760 name field if CID number exists without a CID name. This change improves
761 compatibility with certain device features such as Avaya IP500's directory
764 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
765 created using that setting to not be removed during SIP reload.
767 * Added settings recordonfeature and recordofffeature. When receiving an INFO
768 request with a "Record:" header, this will turn the requested feature on/off.
769 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
770 dynamic features must be enabled and configured properly on the requesting
771 channel for this to function properly.
773 * Add support to realtime for the 'callbackextension' option.
775 * When multiple peers exist with the same address, but differing
776 callbackextension options, incoming requests that are matched by address
777 will be matched to the peer with the matching callbackextension if it is
780 * Two new NAT options, auto_force_rport and auto_comedia, have been added
781 which set the force_rport and comedia options automatically if Asterisk
782 detects that an incoming SIP request crossed a NAT after being sent by
785 * The default global nat setting in sip.conf has been changed from force_rport
788 * NAT settings are now a combinable list of options. The equivalent of the
789 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
791 * Adds an option send_diversion which can be disabled to prevent
792 diversion headers from automatically being added to INVITE requests.
794 * Add support for lightweight NAT keepalive. If enabled a blank packet will
795 be sent to the remote host at a given interval to keep the NAT mapping open.
796 This can be enabled using the keepalive configuration option.
798 * Add option 'tonezone' to specify country code for indications. This option
799 can be set both globally and overridden for specific peers.
801 * The SIP Security Events Framework now supports IPv6.
803 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
804 between multiple user agents. When set, for directmedia reinvites,
805 Asterisk will not send an immediate reinvite on an incoming call leg. This
806 option is useful when peered with another SIP user agent that is known to
807 send immediate direct media reinvites upon call establishment.
809 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
812 * Add options subminexpiry and submaxexpiry to set limits of subscription
813 timer independently from registration timer settings. The setting of the
814 registration timer limits still is done by options minexpiry, maxexpiry
815 and defaultexpiry. For backwards compatibility the setting of minexpiry
816 and maxexpiry also is used to configure the subscription timer limits if
817 subminexpiry and submaxexpiry are not set in sip.conf.
819 * Set registration timer limits to default values when reloading sip
820 configuration and values are not set by configuration.
822 * Add options namedcallgroup and namedpickupgroup to support installations
823 where a higher number of groups (>64) is required.
825 * When a MESSAGE request is received, the address the request was received from
826 is now saved in the SIP_RECVADDR variable.
828 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
829 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
830 the ANI2/OLI information is set on the channel, which can be retrieved using
831 the CALLERID function.
833 * Peers can now be configured to support negotiation of ICE candidates using
834 the setting icesupport. See res_rtp_asterisk changes for more information.
836 * Added support for format attribute negotiation. See the Codecs changes for
839 * Extra headers specified with SIPAddHeader are sent with the REFER message
840 when using Transfer application. See refer_addheaders in sip.conf.sample.
842 * Added support to use private party ID information with calls.
844 * Adds an option discard_remote_hold_retrieval that when set stops telling
845 the peer to start music on hold.
850 * Added skinny version 17 protocol support.
855 * Added ability to use multiple lines for a single phone. This allows multiple
856 calls to occur on a single phone, using callwaiting and switching between calls.
858 * Added option 'sharpdial' allowing end dialing by pressing # key
860 * Added option 'interdigit_timer' to control phone dial timeout
862 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
864 * Added global 'debug' option, that enables debug in channel driver
866 * Added ability to translate on-screen menu in multiple languages. Tested on
867 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
868 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
871 * In addition to English added French and Russian languages for on-screen menus
873 * Reworked dialing number input: added dialing by timeout, immediate dial on
874 on dialplan compare, phone number length now not limited by screen size
876 * Added ability to pickup a call using features.conf defined value and
882 * Add options namedcallgroup and namedpickupgroup to support installations
883 where a higher number of groups (>64) is required.
885 * Added support to use private party ID information with calls.
890 * The minimum DTMF duration can now be configured in asterisk.conf
891 as "mindtmfduration". The default value is (as before) set to 80 ms.
892 (previously it was only available in source code)
894 * Named ACLs can now be specified in acl.conf and used in configurations that
895 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
896 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
897 working ACL. In addition, some CLI commands have been added to provide
898 show information and allow for module reloading - see CLI Changes.
900 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
901 items (separated by commas), and items in the rule can be negated by prefixing
902 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
903 longer necessray to control the order that the 'permit' and 'deny' columns are
904 returned from queries.
906 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
907 be used within the dynamic weight attribute when specifying a mapping.
909 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
910 header, instead of putting the user defined event name there. When enabled
911 the UserDefType header is added for user defined events. This feature is
912 enabled with the setting show_user_defined.
914 * Macro has been deprecated in favor of GoSub. For redirecting and connected
915 line purposes use the following variables instead of their macro equivalents:
916 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
917 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
918 cc_callback_macro in channel configurations.
920 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
923 * Call files now support the "early_media" option to connect with an outgoing
924 extension when early media is received.
926 * Added support to use private party ID information with calls.
931 * A new channel variable, AGIEXITONHANGUP, has been added which allows
932 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
933 AGI application would exit immediately after a channel hangup is detected.
935 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
936 are resolved and each address is attempted in turn until one succeeds or
940 AMI (Asterisk Manager Interface)
942 * The originate action now has an option "EarlyMedia" that enables the
943 call to bridge when we get early media in the call. Previously,
944 early media was disregarded always when originating calls using AMI.
946 * Added setvar= option to manager accounts (much like sip.conf)
948 * Originate now generates an error response if the extension given is not found
951 * MixMonitor will now show IDs associated with the mixmonitor upon creating
952 them if the i(variable) option is used. StopMixMonitor will accept
953 MixMonitorID as an option to close specific MixMonitors.
955 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
956 updated to include information about peers configured with
957 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
958 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
959 returned if auto_force_rport is not enabled.
961 * Added SIPpeerstatus manager command which will generate PeerStatus events
962 similar to the existing PeerStatus events found in chan_sip on demand.
964 * Hangup now can take a regular expression as the Channel option. If you want
965 to hangup multiple channels, use /regex/ as the Channel option. Existing
966 behavior to hanging up a single channel is unchanged, but if you pass a regex,
967 the manager will send you a list of channels back that were hung up.
969 * Support for IPv6 addresses has been added.
971 * AMI Events can now be documented in the Asterisk source. Note that AMI event
972 documentation is only generated when Asterisk is compiled using 'make full'.
973 See the CLI section for commands to display AMI event information.
975 * The AMI Hangup event now includes the AccountCode header so you can easily
976 correlate with AMI Newchannel events.
978 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
979 the StateInterface of the queue member.
981 * Added AMI event SessionTimeout in the Call category that is issued when a
982 call is terminated due to either RTP stream inactivity or SIP session timer
985 * CEL events can now contain a user defined header UserDefType. See core
986 changes for more information.
988 * OOH323 ChannelUpdate events now contain a CallRef header.
990 * Added PresenceState command. This command will report the presence state for
991 the given presence provider.
993 * Added Parkinglots command. This will list all parking lots as a series of
994 AMI Parkinglot events.
996 * Added MessageSend command. This behaves in the same manner as the
997 MessageSend application, and is a technolgoy agnostic mechanism to send out
998 of call text messages.
1000 * Added "message" class authorization. This grants an account permission to
1001 send out of call messages. Write-only.
1006 * The "dialplan add include" command has been modified to create context a context
1007 if one does not already exist. For instance, "dialplan add include foo into bar"
1008 will create context "bar" if it does not already exist.
1010 * A "dialplan remove context" command has been added to remove a context from
1013 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
1014 filenames of all running mixmonitors on a channel.
1016 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
1017 numeric instead of 0, 1, or 2.
1019 * "stun show status" will show a table describing how the STUN client is
1022 * "acl show [named acl]" will show information regarding a Named ACL. The
1023 acl module can be reloaded with "reload acl".
1025 * Added CLI command to display AMI event information - "manager show events",
1026 which shows a list of all known and documented AMI events, and "manager show
1027 event [event name]", which shows detail information about a specific AMI
1030 * The result of the CLI command "queue show" now includes the state interface
1031 information of the queue member.
1033 * The command "core set verbose" will now set a separate level of logging for
1034 each remote console without affecting any other console.
1036 * Added command "cdr show pgsql status" to check connection status
1038 * "sip show channel" will now display the complete route set.
1040 * Added "presencestate list" command. This command will list all custom
1041 presence states that have been set by using the PRESENCE_STATE dialplan
1044 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
1045 command. This changes a custom presence to a new state.
1050 * Codec lists may now be modified by the '!' character, to allow succinct
1051 specification of a list of codecs allowed and disallowed, without the
1052 requirement to use two different keywords. For example, to specify all
1053 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
1055 * Add support for parsing SDP attributes, generating SDP attributes, and
1056 passing it through. This support includes codecs such as H.263, H.264, SILK,
1057 and CELT. You are able to set up a call and have attribute information pass.
1058 This should help considerably with video calls.
1060 * The iLBC codec can now use a system-provided iLBC library if one is installed,
1061 just like the GSM codec.
1065 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
1066 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
1070 * Asterisk version and build information is now logged at the beginning of a
1073 * Threads belonging to a particular call are now linked with callids which get
1074 added to any log messages produced by those threads. Log messages can now be
1075 easily identified as involved with a certain call by looking at their call id.
1076 Call ids may also be attached to log messages for just about any case where
1077 it can be determined to be related to a particular call.
1079 * Each logging destination and console now have an independent notion of the
1080 current verbosity level. Logger.conf now allows an optional argument to
1081 the 'verbose' specifier, indicating the level of verbosity sent to that
1082 particular logging destination. Additionally, remote consoles now each
1083 have their own verbosity level. The command 'core set verbose' will now set
1084 a separate level for each remote console without affecting any other
1090 * Added 'announcement' option which will play at the start of MOH and between
1091 songs in modes of MOH that can detect transitions between songs (eg.
1097 * New per parking lot options: comebackcontext and comebackdialtime. See
1098 configs/features.conf.sample for more details.
1100 * Channel variable PARKER is now set when comebacktoorigin is disabled in
1103 * Channel variable PARKEDCALL is now set with the name of the parking lot
1104 when a timeout occurs.
1110 CDR Postgresql Driver
1112 * Added command "cdr show pgsql status" to check connection status
1115 CDR Adaptive ODBC Driver
1117 * Added schema option for databases that support specifying a schema.
1125 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
1126 CALENDAR_WRITE has completed successfully.
1131 * A new option, 'probation' has been added to rtp.conf
1132 RTP in strictrtp mode can now require more than 1 packet to exit learning
1133 mode with a new source (and by default requires 4). The probation option
1134 allows the user to change the required number of packets in sequence to any
1135 desired value. Use a value of 1 to essentially restore the old behavior.
1136 Also, with strictrtp on, Asterisk will now drop all packets until learning
1137 mode has successfully exited. These changes are based on how pjmedia handles
1138 media sources and source changes.
1140 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
1141 enabled or disabled using the icesupport setting. A variety of other
1142 settings have been introduced to configure STUN/TURN connections.
1147 * A new module, res_corosync, has been introduced. This module uses the
1148 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
1149 of Asterisk servers to both Message Waiting Indication (MWI) and/or
1150 Device State (presence) information. This module is very similar to, and
1151 is a replacement for the res_ais module that was in previous releases of
1157 * This module adds a cleaned up, drop-in replacement for res_jabber called
1158 res_xmpp. This provides the same externally facing functionality but is
1159 implemented differently internally. res_jabber has been deprecated in favor
1160 of res_xmpp; please see the UPGRADE.txt file for more information.
1165 * The safe_asterisk script has been updated to allow several of its parameters
1166 to be set from environment variables. This also enables a custom run
1167 directory of Asterisk to be specified, instead of defaulting to /tmp.
1169 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
1170 its value to determine the directory to assume is the top-level directory of
1171 the source tree. If the variable is not set, it defaults to the current
1172 behavior and uses the current working directory.
1174 ------------------------------------------------------------------------------
1175 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
1176 ------------------------------------------------------------------------------
1180 * Asterisk now has protocol independent support for processing text messages
1181 outside of a call. Messages are routed through the Asterisk dialplan.
1182 SIP MESSAGE and XMPP are currently supported. There are options in
1183 jabber.conf and sip.conf to allow enabling these features.
1184 -> jabber.conf: see the "sendtodialplan" and "context" options.
1185 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
1186 and "outofcall_message_context" options.
1187 The MESSAGE() dialplan function and MessageSend() application have been
1188 added to go along with this functionality. More detailed usage information
1189 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
1190 * If real-time text support (T.140) is negotiated, it will be preferred for
1191 sending text via the SendText application. For example, via SIP, messages
1192 that were once sent via the SIP MESSAGE request would be sent via RTP if
1193 T.140 text is negotiated for a call.
1197 * parkedmusicclass can now be set for non-default parking lots.
1199 Asterisk Manager Interface
1200 --------------------------
1201 * PeerStatus now includes Address and Port.
1202 * Added Hold events for when the remote party puts the call on and off hold
1203 for chan_dahdi ISDN channels.
1204 * Added new action MeetmeListRooms to list active conferences (shows same
1205 data as "meetme list" at the CLI).
1206 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
1207 Description field that is set by 'description' in the channel configuration
1209 * Added Uniqueid header to UserEvent.
1210 * Added new action FilterAdd to control event filters for the current session.
1211 This requires the system permission and uses the same filter syntax as
1212 filters that can be defined in manager.conf
1213 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
1214 versions had some instances of the event converted, but others were left
1215 as-is. All Unlink events should now be converted to Bridge events. The AMI
1216 protocol version number was incremented to 1.2 as a result of this change.
1218 Asterisk HTTP Server
1219 --------------------------
1220 * The HTTP Server can bind to IPv6 addresses.
1223 --------------------------
1224 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
1225 with busydetect. usage example: busypattern=200,200,200,600
1228 --------------------------
1229 * New 'gtalk show settings' command showing the current settings loaded from
1231 * The 'logger reload' command now supports an optional argument, specifying an
1232 alternate configuration file to use.
1233 * 'dialplan add extension' command will now automatically create a context if
1234 the specified context does not exist with a message indicated it did so.
1235 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
1236 Description field which can be populated with 'description' in the channel
1237 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
1240 --------------------------
1241 * The filter option in cdr_adaptive_odbc now supports negating the argument,
1242 thus allowing records which do NOT match the specified filter.
1243 * Added ability to log CONGESTION calls to CDR
1246 --------------------------
1247 * Ability to define custom SILK formats in codecs.conf.
1248 * Addition of speex32 audio format with translation.
1249 * CELT codec pass-through support and ability to define
1250 custom CELT formats in codecs.conf.
1251 * Ability to read raw signed linear files with sample rates
1252 ranging from 8khz - 192khz. The new file extensions introduced
1253 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
1254 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
1255 Skinny, H.323, etc) can still only support the following codecs:
1256 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
1257 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
1258 Video: h261, h263, h263p, h264, mpeg4
1263 --------------------------
1264 * New highly optimized and customizable ConfBridge application capable of
1265 mixing audio at sample rates ranging from 8khz-96khz.
1266 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
1267 and bridge profiles on a channel.
1268 * CONFBRIDGE_INFO dialplan function capable of retrieving information
1269 about a conference such as locked status and number of parties, admins,
1271 * Addition of video_mode option in confbridge.conf for adding video support
1272 into a bridge profile.
1273 * Addition of the follow_talker video_mode in confbridge.conf. This video
1274 mode dynamically switches the video feed to always display the loudest talker
1275 supplying video in the conference.
1279 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
1280 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
1281 variables from asterisk.conf.
1285 * Addition of the JITTERBUFFER dialplan function. This function allows
1286 for jitterbuffering to occur on the read side of a channel. By using
1287 this function conference applications such as ConfBridge and MeetMe can
1288 have the rx streams jitterbuffered before conference mixing occurs.
1289 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
1291 * Added STRREPLACE function. This function let's the user search a variable
1292 for a given string to replace with another string as many times as the
1293 user specifies or just throughout the whole string.
1294 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
1295 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
1296 * Added extensions to chan_ooh323 in function CHANNEL()
1298 libpri channel driver (chan_dahdi) DAHDI changes
1299 --------------------------
1300 * Added moh_signaling option to specify what to do when the channel's bridged
1301 peer puts the ISDN channel on hold.
1302 * Added display_send and display_receive options to control how the display ie
1303 is handled. To send display text from the dialplan use the SendText()
1304 application when the option is enabled.
1305 * Added mcid_send option to allow sending a MCID request on a span.
1308 --------------------------
1309 * Added setvar option to calendar.conf to allow setting channel variables on
1310 notification channels.
1311 * Added "calendar show types" CLI command to list registered calendar
1315 --------------------------
1316 * Added two new options, r and t with file name arguments to record
1317 single direction (unmixed) audio recording separate from the bidirectional
1318 (mixed) recording. The mixed file name argument is optional now as long
1319 as at least one recording option is used.
1322 --------------------------
1323 * Added a new option, l, which will disable local call optimization for
1324 channels involved with the FollowMe thread. Use this option to improve
1325 compatability for a FollowMe call with certain dialplan apps, options, and
1329 --------------------------
1330 * Added option "k" that will automatically close the conference when there's
1331 only one person left when a user exits the conference.
1334 --------------------------
1335 * cel_pgsql now supports the 'extra' column for data added using the
1336 CELGenUserEvent() application.
1339 --------------------------
1340 * Support for defining hints has been added to pbx_lua. See the 'hints' table
1341 in the sample extensions.lua file for syntax details.
1342 * Applications that perform jumps in the dialplan such as Goto will now
1343 execute properly. When pbx_lua detects that the context, extension, or
1344 priority we are executing on has changed it will immediately return control
1345 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
1346 the priority after the currently executing priority.
1347 * An autoservice is now started by default for pbx_lua channels. It can be
1348 stopped and restarted using the autoservice_stop() and autoservice_start()
1352 --------------------------
1353 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
1354 into a FAXStatus event with an 'Operation' header that will be either
1355 'send', 'receive', and 'gateway'.
1356 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
1357 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
1358 feature will handle converting a fax call between an audio T.30 fax terminal
1359 and an IFP T.38 fax terminal.
1363 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
1364 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
1365 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
1369 * Added general option negative_penalty_invalid default off. when set
1370 members are seen as invalid/logged out when there penalty is negative.
1371 for realtime members when set remove from queue will set penalty to -1.
1372 * Added queue option autopausedelay when autopause is enabled it will be
1373 delayed for this number of seconds since last successful call if there
1374 was no prior call the agent will be autopaused immediately.
1375 * Added member option ignorebusy this when set and ringinuse is not
1376 will allow per member control of multiple calls as ringinuse does for
1381 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
1383 * Added 'k' option to MeetMe to automatically kill the conference when there's only
1384 one participant left (much like a normal call bridge)
1385 * Added extra argument to Originate to set timeout.
1389 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
1390 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
1391 utility in the UTILS section of menuselect. If an existing astdb is found and no
1392 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
1393 convert an existing astdb to the SQLite3 version automatically at runtime.
1397 * Modules marked as deprecated are no longer marked as building by default. Enabling
1398 these modules is still available via menuselect.
1402 * authdebug is now disabled by default. To enable this functionaility again
1403 set authdebug = yes in iax.conf.
1407 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
1408 releases it was disabled.
1412 * The PBX core previously made a call with a non-existing extension test for
1413 extension s@default and jump there if the extension existed.
1414 This was a bad default behaviour and violated the principle of least surprise.
1415 It has therefore been changed in this release. It may affect some
1416 applications and configurations that rely on this behaviour. Most channel
1417 drivers have avoided this for many releases by testing whether the extension
1418 called exists before starting the PBX and generating a local error.
1419 This behaviour still exists and works as before.
1421 Extension "s" is used when no extension is given in a channel driver,
1422 like immediate answer in DAHDI or calling to a domain with no user part
1425 ------------------------------------------------------------------------------
1426 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
1427 ------------------------------------------------------------------------------
1431 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
1432 now defaults to force_rport. It is very important that phones requiring nat=no be
1433 specifically set as such instead of relying on the default setting. If at all
1434 possible, all devices should have nat settings configured in the general section as
1435 opposed to configuring nat per-device.
1436 * Added preferred_codec_only option in sip.conf. This feature limits the joint
1437 codecs sent in response to an INVITE to the single most preferred codec.
1438 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
1439 to be used for the outgoing call. It must be one of the codecs configured
1441 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
1442 to be used for holding a private key. If tlsprivatekey is not specified,
1443 tlscertfile is searched for both public and private key.
1444 * Added tlsclientmethod option to sip.conf. This allows the protocol for
1445 outbound client connections to be specified.
1446 * The sendrpid parameter has been expanded to include the options
1447 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
1448 header to be sent (equivalent to setting sendrpid=yes) and setting
1449 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
1450 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
1451 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
1452 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
1453 will accept the SDP even if the SDP version number is not properly incremented,
1454 but will generate a warning in the log indicating that the SIP peer that sent
1455 the SDP should have the 'ignoresdpversion' option set.
1456 * The 'nat' option has now been been changed to have yes, no, force_rport, and
1457 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
1458 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
1459 remote side requests it and disables symmetric RTP support. Setting it to
1460 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
1461 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
1462 and enables symmetric RTP support.
1463 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
1464 response. This permits the master channel to know how each channel dialled
1465 in a multi-channel setup resolved in an individual way. This carries a
1466 performance penalty and can be disabled in sip.conf using the
1467 'storesipcause' option.
1468 * Added 'externtcpport' and 'externtlsport' options to allow custom port
1469 configuration for the externip and externhost options when tcp or tls is used.
1470 * Added support for message body (stored in content variable) to SIP NOTIFY message
1471 accessible via AMI and CLI.
1472 * Added 'media_address' configuration option which can be used to explicitly specify
1473 the IP address to use in the SDP for media (audio, video, and text) streams.
1474 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
1475 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
1477 * Added 'use_q850_reason' configuration option for generating and parsing
1478 if available Reason: Q.850;cause=<cause code> header. It is implemented
1479 in some gateways for better passing PRI/SS7 cause codes via SIP.
1480 * When dialing SIP peers, a new component may be added to the end of the dialstring
1481 to indicate that a specific remote IP address or host should be used when dialing
1482 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
1483 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
1484 ability to selectively force bridged channels to also be encrypted is also
1485 implemented. Branching in the dialplan can be done based on whether or not
1486 a channel has secure media and/or signaling.
1487 * Added directmediapermit/directmediadeny to limit which peers can send direct media
1489 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
1490 Charge messages to snom phones.
1491 * Added support for G.719 media streams.
1492 * Added support for 16khz signed linear media streams.
1493 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
1494 RTP has been outfitted with the same abilities.
1495 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
1496 available in device configurations as well as in the dial plan.
1497 * Addition of the 'subscribe_network_change' option for turning on and off
1498 res_stun_monitor module support in chan_sip.
1499 * Addition of the 'auth_options_requests' option for turning on and off
1500 authentication for OPTIONS requests in chan_sip.
1504 * Add #tryinclude statement for config files. This provides the same
1505 functionality as the #include statement however an asterisk module will
1506 still load if the filename does not exist. Using the #include statement
1507 Asterisk will not allow the module to load.
1511 * Added rtsavesysname option into iax.conf to allow the systname to be saved
1512 on realtime updates.
1513 * Added the ability for chan_iax2 to inform the dialplan whether or not
1514 encryption is being used. This interoperates with the SIP SRTP implementation
1515 so that a secure SIP call can be bridged to a secure IAX call when the
1516 dialplan requires bridged channels to be "secure".
1517 * Addition of the 'subscribe_network_change' option for turning on and off
1518 res_stun_monitor module support in chan_iax.
1523 * Added ability to preset channel variables on indicated lines with the setvar
1524 configuration option. Also, clearvars=all resets the list of variables back
1526 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
1527 See configs/res_pktccops.conf for more information.
1529 XMPP Google Talk/Jingle changes
1530 -------------------------------
1531 * Added the externip option to gtalk.conf.
1532 * Added the stunaddr option to gtalk.conf which allows for the automatic
1533 retrieval of the external ip from a stun server.
1537 * Added 'p' option to PickupChan() to allow for picking up channel by the first
1538 match to a partial channel name.
1539 * Added .m3u support for Mp3Player application.
1540 * Added progress option to the app_dial D() option. When progress DTMF is
1541 present, those values are sent immediately upon receiving a PROGRESS message
1542 regardless if the call has been answered or not.
1543 * Added functionality to the app_dial F() option to continue with execution
1544 at the current location when no parameters are provided.
1545 * Added the 'a' option to app_dial to answer the calling channel before any
1546 announcements or macros are executed.
1547 * Modified app_dial to set answertime when the called channel answers even if
1548 the called channel hangs up during playback of an announcement.
1549 * Modified app_dial 'r' option to support an additional parameter to play an
1550 indication tone from indications.conf
1551 * Added c() option to app_chanspy. This option allows custom DTMF to be set
1552 to cycle through the next available channel. By default this is still '*'.
1553 * Added x() option to app_chanspy. This option allows DTMF to be set to
1554 exit the application.
1555 * The Voicemail application has been improved to automatically ignore messages
1556 that only contain silence.
1557 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
1558 associated mailbox(es) to be greetings-only.
1559 * The ChanSpy application now has the 'S' option, which makes the application
1560 automatically exit once it hits a point where no more channels are available
1562 * The ChanSpy application also now has the 'E' option, which spies on a single
1563 channel and exits when that channel hangs up.
1564 * The MeetMe application now turns on the DENOISE() function by default, for
1565 each participant. In our tests, this has significantly decreased background
1566 noise (especially noisy data centers).
1567 * Voicemail now permits storage of secrets in a separate file, located in the
1568 spool directory of each individual user. The control for this is located in
1569 the "passwordlocation" option in voicemail.conf. Please see the sample
1570 configuration for more information.
1571 * The ChanIsAvail application now exposes the returned cause code using a separate
1572 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
1573 * Added 'd' option to app_followme. This option disables the "Please hold"
1575 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
1576 received will terminate recording.
1577 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
1578 Previously the folder could only be set per context, but has now been extended
1579 using the imapfolder option.
1580 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
1581 * Voicemail now allows the pager date format to be specified separately from the
1583 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
1584 to allow joining, leaving, and sending text to group chats.
1585 * MeetMe has a new option 'G' to play an announcement before joining a conference.
1586 * Page has a new option 'A(x)' which will playback an announcement simultaneously
1587 to all paged phones (and optionally excluding the caller's one using the new
1588 option 'n') before the call is bridged.
1589 * The 'f' option to Dial has been augmented to take an optional argument. If no
1590 argument is provided, the 'f' option works as it always has. If an argument is
1591 provided, then the connected party information of all outgoing channels created
1592 during the Dial will be set to the argument passed to the 'f' option.
1593 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
1595 * The OSP lookup application adds in/outbound network ID, optional security,
1596 number portability, QoS reporting, destination IP port, custom info and service
1598 * Added new application VMSayName that will play the recorded name of the voicemail
1599 user if it exists, otherwise will play the mailbox number.
1600 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
1601 retrieve state for a particular bridge, where <name> is the conference name
1602 * app_directory now allows exiting at any time using the operator or pound key.
1603 * Voicemail now supports setting a locale per-mailbox.
1604 * Two new applications are provided for declining counting phrases in multiple
1605 languages. See the application notes for SayCountedNoun and SayCountedAdj for
1607 * Voicemail now runs the externnotify script when pollmailboxes is activated and
1609 * Voicemail now includes rdnis within msgXXXX.txt file.
1610 * ExternalIVR now supports IPv6 addresses.
1611 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
1612 at https://wiki.asterisk.org/wiki/x/oQBB
1613 * ParkedCall and Park can now specify the parking lot to use.
1617 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
1618 over SRV records associated with a specific service. From the CLI, type
1619 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
1620 details on how these may be used.
1621 * PITCH_SHIFT dialplan function added. This function can be used to modify the
1622 pitch of a channel's tx and rx audio streams.
1623 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
1624 setting various connected line and redirecting party information.
1625 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
1626 support ISDN subaddressing.
1627 * The CHANNEL() function now supports the "name" and "checkhangup" options.
1628 * For DAHDI channels, the CHANNEL() dialplan function now allows
1629 the dialplan to request changes in the configuration of the active
1630 echo canceller on the channel (if any), for the current call only.
1633 exten => s,n,Set(CHANNEL(echocan_mode)=off)
1635 The possible values are:
1637 on - normal mode (the echo canceller is actually reinitialized)
1639 fax - FAX/data mode (NLP disabled if possible, otherwise completely
1641 voice - voice mode (returns from FAX mode, reverting the changes that
1642 were made when FAX mode was requested)
1643 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
1644 and setting variables on the channel which created the current channel.
1645 Administrators should take care to avoid naming conflicts, when multiple
1646 channels are dialled at once, especially when used with the Local channel
1647 construct (which all could set variables on the master channel). Usage
1648 of the HASH() dialplan function, with the key set to the name of the slave
1649 channel, is one approach that will avoid conflicts.
1650 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
1652 * func_odbc now allows multiple row results to be retrieved without using
1653 mode=multirow. If rowlimit is set, then additional rows may be retrieved
1654 from the same query by using the name of the function which retrieved the
1655 first row as an argument to ODBC_FETCH().
1656 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
1657 dialplan. This function returns the content of the received message.
1658 * Added REPLACE, which searches a given variable name for a set of characters,
1659 then either replaces them with a single character or deletes them.
1660 * Added PASSTHRU, which literally passes the same argument back as its return
1661 value. The intent is to be able to use a literal string argument to
1662 functions that currently require a variable name as an argument.
1663 * HASH-associated variables now can be inherited across channel creation, by
1664 prefixing the name of the hash at assignment with the appropriate number of
1665 underscores, just like variables.
1666 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
1667 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
1668 whether or not channels that are bridged to the current channel will be
1669 required to have secure signaling and/or media.
1670 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
1671 the current channel has secure signaling and/or media.
1672 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
1673 "no_media_path" option.
1674 Returns "0" if there is a B channel associated with the call.
1675 Returns "1" if no B channel is associated with the call. The call is either
1676 on hold or is a call waiting call.
1677 * Added option to dialplan function CDR(), the 'f' option
1678 allows for high resolution times for billsec and duration fields.
1679 * FILE() now supports line-mode and writing.
1680 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
1681 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
1685 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
1686 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
1687 and is set when a dynamic feature is triggered.
1688 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
1689 to dynamically create a new parking lot matching the value this varible is
1691 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
1692 features.conf that should be the base for dynamic parkinglots.
1693 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
1694 parkinglot should have.
1695 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
1696 parkinglot should have.
1697 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
1702 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
1703 timeout has expired.
1704 * Added 'R' option to app_queue. This option stops moh and indicates ringing
1705 to the caller when an Agent's phone is ringing. This can be used to indicate
1706 to the caller that their call is about to be picked up, which is nice when
1707 one has been on hold for an extened period of time.
1708 * A new config option, penaltymemberslimit, has been added to queues.conf.
1709 When set this option will disregard penalty settings when a queue has too
1711 * A new option, 'I' has been added to both app_queue and app_dial.
1712 By setting this option, Asterisk will not update the caller with
1713 connected line changes or redirecting party changes when they occur.
1714 * A 'relative-periodic-announce' option has been added to queues.conf. When
1715 enabled, this option will cause periodic announce times to be calculated
1716 from the end of announcements rather than from the beginning.
1717 * The autopause option in queues.conf can be passed a new value, "all." The
1718 result is that if a member becomes auto-paused, he will be paused in all
1719 queues for which he is a member, not just the queue that failed to reach
1721 * Added dialplan function QUEUE_EXISTS to check if a queue exists
1722 * The queue logger now allows events to optionally propagate to a file,
1723 even when realtime logging is turned on. Additionally, realtime logging
1724 supports sending the event arguments to 5 individual fields, although it
1725 will fallback to the previous data definition, if the new table layout is
1728 mISDN channel driver (chan_misdn) changes
1729 ----------------------------------------
1730 * Added display_connected parameter to misdn.conf to put a display string
1731 in the CONNECT message containing the connected name and/or number if
1732 the presentation setting permits it.
1733 * Added display_setup parameter to misdn.conf to put a display string
1734 in the SETUP message containing the caller name and/or number if the
1735 presentation setting permits it.
1736 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
1737 indicate the dialplan settings are to be obtained from the asterisk
1739 * Made misdn.conf parameter callerid accept the "name" <number> format
1740 used by the rest of the system.
1741 * Made use the nationalprefix and internationalprefix misdn.conf
1742 parameters to prefix any received number from the ISDN link if that
1743 number has the corresponding Type-Of-Number. NOTE: This includes
1744 comparing the incoming call's dialed number against the MSN list.
1745 * Added the following new parameters: unknownprefix, netspecificprefix,
1746 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
1747 received number from the ISDN link if that number has the corresponding
1749 * Added new dialplan application misdn_command which permits controlling
1750 the CCBS/CCNR functionality.
1751 * Added new dialplan function mISDN_CC which permits retrieval of various
1752 values from an active call completion record.
1753 * For PTP, you should manually send the COLR of the redirected-to party
1754 for an incomming redirected call if the incoming call could experience
1755 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
1756 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
1757 if the REDIRECTING(from-num) is not empty.
1758 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
1759 option on all of the REDIRECTING statements before dialing the
1760 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
1761 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
1762 redirecting-to presentation (COLR) when it becomes available.
1763 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
1766 thirdparty mISDN enhancements
1767 -----------------------------
1768 mISDN has been modified by Digium, Inc. to greatly expand facility message
1770 * Enhanced COLP support for call diversion and transfer.
1771 * CCBS/CCNR support.
1773 The latest modified mISDN v1.1.x based version is available at:
1774 http://svn.digium.com/svn/thirdparty/mISDN/trunk
1775 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
1777 Tagged versions of the modified mISDN code are available under:
1778 http://svn.digium.com/svn/thirdparty/mISDN/tags
1779 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
1781 libpri channel driver (chan_dahdi) DAHDI changes
1782 -------------------------------------------
1783 * The channel variable PRIREDIRECTREASON is now just a status variable
1784 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
1785 to read and alter the reason.
1786 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
1787 redirected-to party for an incomming redirected call if the incoming call
1788 could experience further redirects. Just set the
1789 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
1790 to the COLR. A call has been redirected if the REDIRECTING(count) is not
1792 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
1793 use the inhibit(i) option on all of the REDIRECTING statements before
1794 dialing the redirected-to party. You still have to set the
1795 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
1796 will update the redirecting-to presentation (COLR) when it becomes available.
1797 * Added the ability to ignore calls that are not in a Multiple Subscriber
1798 Number (MSN) list for PTMP CPE interfaces.
1799 * Added dynamic range compression support for dahdi channels. It is
1800 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
1801 * Added support for ISDN calling and called subaddress with partial support
1802 for connected line subaddress.
1803 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
1804 * Added handling of received HOLD/RETRIEVE messages and the optional ability
1805 to transfer a held call on disconnect similar to an analog phone.
1806 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
1807 Will reroute/deflect an outgoing call when receive the message.
1808 Can use the DAHDISendCallreroutingFacility to send the message for the
1810 * Added standard location to add options to chan_dahdi dialing:
1811 Dial(DAHDI/g1[/extension[/options]])
1814 R Reverse charging indication
1815 * Added Reverse Charging Indication (Collect calls) send/receive option.
1816 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
1817 Dial(DAHDI/g1/extension/R)
1818 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
1819 (requires latest LibPRI)
1820 * Added ability to send/receive keypad digits in the SETUP message.
1821 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
1822 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
1823 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
1824 (requires latest LibPRI)
1825 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
1826 to eliminate tromboned calls. A tromboned call goes out an interface and comes
1827 back into the same interface. Tromboned calls happen because of call routing,
1828 call deflection, call forwarding, and call transfer.
1829 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
1830 * Added the ability to support call waiting calls. (The SETUP has no B channel
1832 * Added Malicious Call ID (MCID) event to the AMI call event class.
1833 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
1835 Asterisk Manager Interface
1836 --------------------------
1837 * The Hangup action now accepts a Cause header which may be used to
1838 set the channel's hangup cause.
1839 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
1840 to specify a separate .pem file to hold a private key. By default sslcert
1841 is used to hold both the public and private key.
1842 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
1843 for options containing the 'tls' prefix. For example, 'sslenable' is now
1844 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
1845 across all .conf files. All affected sample.conf files have been modified to
1846 reflect this change. Previous options such as 'sslenable' still work,
1847 but options with the 'tls' prefix are preferred.
1848 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
1849 in a channel. (res_mutestream.so)
1850 * The configuration file manager.conf now supports a channelvars option, which
1851 specifies a list of channel variables to include in each channel-oriented
1853 * The redirect command now has new parameters ExtraContext, ExtraExtension,
1854 and ExtraPriority to allow redirecting the second channel to a different
1855 location than the first.
1856 * Added new event "JabberStatus" in the Jabber module to monitor buddies
1858 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
1859 in a MixMonitor recording.
1860 * The 'iax2 show peers' output is now similar to the expected output of
1862 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
1864 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
1865 AOC-E messages on a channel.
1866 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
1867 conform more closely to similar events.
1868 * Added a new eventfilter option per user to allow whitelisting and blacklisting
1870 * Added optional parkinglot variable for park command.
1871 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
1872 if CallerIDNum and CallerIDName headers are also present.
1874 Channel Event Logging
1875 ---------------------
1876 * A new interface, CEL, is introduced here. CEL logs single events, much like
1877 the AMI, but it differs from the AMI in that it logs to db backends much
1878 like CDR does; is based on the event subsystem introduced by Russell, and
1879 can share in all its benefits; allows multiple backends to operate like CDR;
1880 is specialized to event data that would be of concern to billing sytems,
1881 like CDR. Backends for logging and accounting calls have been produced,
1882 but a new CDR backend is still in development.
1886 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
1887 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
1888 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
1889 * Multiple files and formats can now be specified in cdr_custom.conf.
1890 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
1891 See configs/cdr_syslog.conf.sample for more information.
1892 * A 'sequence' field has been added to CDRs which can be combined with
1893 linkedid or uniqueid to uniquely identify a CDR.
1894 * Handling of billsec and duration field has changed. If your table definition
1895 specifies those fields as float,double or similar they will now be logged with
1896 microsecond accuracy instead of a whole integer.
1898 Calendaring for Asterisk
1899 ------------------------
1900 * A new set of modules were added supporing calendar integration with Asterisk.
1901 Dialplan functions for reading from and writing to calendars are included,
1902 as well as the ability to execute dialplan logic upon calendar event notifications.
1903 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
1904 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
1905 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
1906 2003 support does not support forms-based authentication).
1908 Call Completion Supplementary Services for Asterisk
1909 ---------------------------------------------------
1910 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
1911 DAHDI/ISDN supports call completion for the following switch types:
1912 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
1913 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
1915 Multicast RTP Support
1916 ---------------------
1917 * A new RTP engine and channel driver have been added which supports Multicast RTP.
1918 The channel driver can be used with the Page application to perform multicast RTP
1919 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
1920 Type can be either basic or linksys.
1921 Destination is the IP address and port for the RTP packets.
1922 Control address is specific to the linksys type and is used for sending the control
1923 packets unique to them.
1925 Security Events Framework
1926 -------------------------
1927 * Asterisk has a new C API for reporting security events. The module res_security_log
1928 sends these events to the "security" logger level. Currently, AMI is the only
1929 Asterisk component that reports security events. However, SIP support will be
1930 coming soon. For more information on the security events framework, see the
1931 "Asterisk Security Framework" section of the Asterisk wiki at
1932 https://wiki.asterisk.org/wiki/x/wgBQ
1933 * SIP support was added in Asterisk 10
1934 * This API now supports IPv6 addresses
1938 * A technology independent fax frontend (res_fax) has been added to Asterisk.
1939 * A spandsp based fax backend (res_fax_spandsp) has been added.
1940 * The app_fax module has been deprecated in favor of the res_fax module and
1941 the new res_fax_spandsp backend.
1942 * The SendFAX and ReceiveFAX applications now send their log messages to a
1943 'fax' logger level, instead of to the generic logger levels. To see these
1944 messages, the system's logger.conf file will need to direct the 'fax' logger
1945 level to one or more destinations; the logger.conf.sample file includes an
1946 example of how to do this. Note that if the 'fax' logger level is *not*
1947 directed to at least one destination, log messages generated by these
1948 applications will be lost, and that if the 'fax' logger level is directed to
1949 the console, the 'core set verbose' and 'core set debug' CLI commands will
1950 have no effect on whether the messages appear on the console or not.
1954 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
1955 Now, in order to enable transmitting silence during record the transmit_silence
1956 option should be used. transmit_silence_during_record remains a valid option, but
1957 defaults to the behavior of the transmit_silence option.
1958 * Addition of the Unit Test Framework API for managing registration and execution
1959 of unit tests with the purpose of verifying the operation of C functions.
1960 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
1961 XMPP text messages to the remote JID.
1962 * Modules.conf has a new option - "require" - that marks a module as critical for
1963 the execution of Asterisk.
1964 If one of the required modules fail to load, Asterisk will exit with a return
1966 * An 'X' option has been added to the asterisk application which enables #exec support.
1967 This allows #exec to be used in asterisk.conf.
1968 * jabber.conf supports a new option auth_policy that toggles auto user registration.
1969 * A new lockconfdir option has been added to asterisk.conf to protect the
1970 configuration directory (/etc/asterisk by default) during reloads.
1971 * The parkeddynamic option has been added to features.conf to enable the creation
1972 of dynamic parkinglots.
1973 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
1974 the reportalarms config option.
1975 * chan_dahdi supports dialing configuring and dialing by device file name.
1976 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
1977 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
1978 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
1979 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
1980 Handy for the above name-based syntax as it does not depend on
1981 initialization order.
1982 * The Realtime dialplan switch now caches entries for 1 second. This provides a
1983 significant increase in performance (about 3X) for installations using this switchtype.
1984 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
1985 AIS. For more information, please see the Distributed Device State section of the
1986 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1987 * The addition of G.719 pass-through support.
1988 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
1989 during device configuration.
1990 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
1991 have less than 3 lines on the LCD.
1992 * Realtime now supports database failover. See the sample extconfig.conf for details.
1993 * The addition of improved translation path building for wideband codecs. Sample
1994 rate changes during translation are now avoided unless absolutely necessary.
1995 * The addition of the res_stun_monitor module for monitoring and reacting to network
1996 changes while behind a NAT.
1997 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
1998 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
1999 These allow support for any Administration. Default is AT&T values.
2003 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
2004 optionally accept a filename, to apply the setting only to the code generated from
2005 that source file when Asterisk was built. However, there are some modules in Asterisk
2006 that are composed of multiple source files, so this did not result in the behavior
2007 that users expected. In this version, 'core set debug' and 'core set verbose'
2008 can optionally accept *module* names instead (with or without the .so extension),
2009 which applies the setting to the entire module specified, regardless of which source
2010 files it was built from.
2011 * New 'manager show settings' command showing the current settings loaded from
2013 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
2014 the channel hangup request to all channels.
2015 * Added a "core reload" CLI command that executes a global reload of Asterisk.
2017 ------------------------------------------------------------------------------
2018 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
2019 ------------------------------------------------------------------------------
2023 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
2024 Snom phones use this for call pickup of extensions that the phone is
2026 * Added support for setting the domain in the URI for caller of an
2027 outbound call by using the SIPFROMDOMAIN channel variable.
2028 * Added a new configuration option "remotesecret" for authentication to
2029 remote services. For backwards compatibility, "secret" still has the
2030 same function as before, but now you can configure both a remote secret and a
2031 local secret for mutual authentication.
2032 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
2033 the sound will be played to the target of an attended transfer
2034 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
2035 finer control over how many peers Asterisk will qualify and the gap between them
2036 when all peers need to be qualified at the same time.
2037 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
2038 (either globally or for a specific peer), chan_sip will treat any SDP data
2039 it receives as new data and update the media stream accordingly. By
2040 default, Asterisk will only modify the media stream if the SDP session
2041 version received is different from the current SDP session version. This
2042 option is required to interoperate with devices that have non-standard SDP
2043 session version implementations (observed with Microsoft OCS). This option
2044 is disabled by default.
2045 * The parsing of register => lines in sip.conf has been modified to allow a port
2046 to be present in the "user" portion. Please see the sip.conf.sample file for more
2048 * Added support for subscribing to MWI on a remote server and making the status available
2049 as a mailbox. Please see the sip.conf.sample file for more information.
2050 * Added a function to remove SIP headers added in the dialplan before the
2051 first INVITE is generated - SIPRemoveHeader()
2052 * Channel variables set with setvar= in a device configuration is now
2053 set both for inbound and outbound calls.
2054 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
2058 * Added immediate option to iax.conf
2059 * Added forceencryption option to iax.conf
2060 * Added Encryption and Trunk status to manager command "iaxpeers"
2064 * The configuration file now holds separate sections for devices and lines.
2065 Please have a look at configs/skinny.conf.sample and change your skinny.conf
2070 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
2071 support for LibOpenR2. http://www.libopenr2.org/
2072 * The UK option waitfordialtone has been added for use with BT analog
2074 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
2075 is used in conjunction with the 'faxdetect' configuration option. When
2076 'faxbuffers' is used and fax tones are detected, the channel will dynamically
2077 switch to the configured faxbuffers policy. For example, to use 6 buffers
2078 and a 'full' buffer policy for a fax transmission, add:
2080 The faxbuffers configuration will be in affect until the call is torn down.
2081 * Added service message support for 4ESS/5ESS switches.
2085 * For DAHDI channels, the CHANNEL() dialplan function now
2086 supports changing the channel's buffer policy (for the current
2087 call only), using this syntax:
2089 exten => s,n,Set(CHANNEL(buffers)=6,full)
2091 This would change the channel to the 'full' buffer policy and
2092 6 (six) buffers. Possible options for this setting are the same
2093 as those in chan_dahdi.conf.
2094 * Added a new dialplan function, CURLOPT, which permits setting various
2095 options that may be useful with the CURL dialplan function, such as
2096 cookies, proxies, connection timeouts, passwords, etc.
2097 * Permit the syntax and synopsis fields of the corresponding dialplan
2098 functions to be individually set from func_odbc.conf.
2099 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
2100 * func_odbc now may specify an insert query to execute, when the write query
2101 affects 0 rows (usually indicating that no such row exists).
2102 * Added a new dialplan function, LISTFILTER, which permits removing elements
2103 from a set list, by name. Uses the same general syntax as the existing CUT
2104 and FIELDQTY dialplan functions, which also manage lists.
2105 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
2106 obtaining realtime data from the dialplan.
2107 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
2108 a subroutine when using the GoSub() and Return() applications.
2109 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
2110 of "core show function AUDIOHOOK_INHERIT" from the CLI
2111 * Added AES_ENCRYPT. For information on its use, please see the output
2112 of "core show function AES_ENCRYPT" from the CLI
2113 * Added AES_DECRYPT. For information on its use, please see the output
2114 of "core show function AES_DECRYPT" from the CLI
2115 * func_odbc now supports database transactions across multiple queries.
2119 * Scheduled meetme conferences may now have their end times extended by
2121 * app_authenticate now gives the ability to select a prompt other than
2123 * app_directory now pays attention to the searchcontexts setting in
2124 voicemail.conf and will look through all contexts, if no context is
2125 specified in the initial argument.
2126 * A new application, Originate, has been introduced, that allows asynchronous
2127 call origination from the dialplan.
2128 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
2129 in addition to the setting in the "general" context.
2130 * Added ConfBridge dialplan application which does conference bridges without
2131 DAHDI. For information on its use, please see the output of
2132 "core show application ConfBridge" from the CLI.
2136 * The Asterisk CLI has a new command, "channel redirect", which is similar in
2137 operation to the AMI Redirect action.
2138 * extensions.conf now allows you to use keyword "same" to define an extension
2139 without actually specifying an extension. It uses exactly the same pattern
2140 as previously used on the last "exten" line. For example:
2141 exten => 123,1,NoOp(something)
2142 same => n,SomethingElse()
2143 * musiconhold.conf classes of type 'files' can now use relative directory paths,
2144 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
2145 * All deprecated CLI commands are removed from the sourcecode. They are now handled
2146 by the new clialiases module. See cli_aliases.conf.sample file.
2147 * Times within timespecs are now accurate down to the minute. This is a change
2148 from historical Asterisk, which only provided timespecs rounded to the nearest
2149 even (read: evenly divisible by 2) minute mark.
2150 * The realtime switch now supports an option flag, 'p', which disables searches for
2152 * In addition to a time range and date range, timespecs now accept a 5th optional
2153 argument, timezone. This allows you to perform time checks on alternate
2154 timezones, especially if those daylight savings time ranges vary from your
2155 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
2157 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
2158 give you the correct output for an asterisk box behind nat. It will give you the
2159 externhost and localnet settings.
2160 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
2161 can connect calls in passthrough mode, as well as record and play back files.
2162 * Successful and unsuccessful call pickup can now be alerted through sounds, by
2163 using pickupsound and pickupfailsound in features.conf.
2164 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
2165 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
2166 instead of the /var/run/asterisk.pid where it used to be. This will make
2167 installs as non-root easier to manage.
2172 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
2173 be written; they will no longer be explicitly written.
2175 Asterisk Manager Interface
2176 --------------------------
2177 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
2178 a non-empty value) in your request. If you do this, any pending AMI events will
2179 *not* be included in the response to your request as they would normally, but
2180 will be left in the event queue for the next request you make to retrieve. For
2181 some applications, this will allow you to guarantee that you will only see
2182 events in responses to 'WaitEvent' actions, and can better know when to expect them.
2183 To know whether the Asterisk server supports this header or not, your client can
2184 inspect the first response back from the server to see if it includes this header:
2186 Pragma: SuppressEvents
2188 If this is included, the server supports event suppression.
2190 * Added 4 new Actions to list skinny device(s) and line(s)
2196 LDAP Schema File Additions
2197 --------------------------
2198 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
2199 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
2201 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
2202 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
2203 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
2204 * Removed redundant IPaddr (there's already IPAddress)
2205 - Gives more configuration Flags for SIP-Users available (tested)
2206 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
2207 without extensibleObject (which really should be the last resort); gives
2208 also additional possibilities for LDAP-filter
2210 ------------------------------------------------------------------------------
2211 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
2212 ------------------------------------------------------------------------------
2214 Device State Handling
2215 ---------------------
2216 * The event infrastructure in Asterisk got another big update to help support
2217 distributed events. It currently supports distributed device state and
2218 distributed Voicemail MWI (Message Waiting Indication). A new module has
2219 been merged, res_ais, which facilitates communicating events between servers.
2220 It uses the SAForum AIS (Service Availability Forum Application Interface
2221 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
2222 a cluster of Asterisk servers, and to share events between them. For more
2223 information on setting this up, refer to the Distributed Device State section
2224 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
2228 * Added a new dialplan function, AST_CONFIG(), which allows you to access
2229 variables from an Asterisk configuration file.
2230 * The JACK_HOOK function now has a c() option to supply a custom client name.
2231 * Added two new dialplan functions from libspeex for audio gain control and
2232 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
2233 rx directions of a channel from the dialplan.
2234 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
2235 based on other parameters. The default is still to search based on the
2236 forwarding station ID. However, there are new options that allow you to search
2237 based on the message desk terminal ID, or the message desk number.
2238 * TIMEOUT() has been modified to be accurate down to the millisecond.
2239 * ENUM*() functions now include the following new options:
2240 - 'u' returns the full URI and does not strip off the URI-scheme.
2241 - 's' triggers ISN specific rewriting
2242 - 'i' looks for branches into an Infrastructure ENUM tree
2243 - 'd' for a direct DNS lookup without any flipping of digits.
2244 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
2245 * CHANNEL() now has options for the maximum, minimum, and standard or normal
2246 deviation of jitter, rtt, and loss for a call using chan_sip.
2248 DAHDI channel driver (chan_dahdi) Changes
2249 ----------------------------------------
2250 * Channels can now be configured using named sections in chan_dahdi.conf, just
2251 like other channel drivers, including the use of templates.
2252 * The default for pridialplan has changed from 'national' to 'unknown'.
2256 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
2257 to something that matches the pattern a hint will be created using the contents
2258 and variables evaluated.
2259 * Dialplan matching has been extended to allow an extension to return to the
2260 PBX core to wait for more digits. This is done by using the new dialplan
2261 application called "Incomplete". This will permit a whole new level of
2262 extension control, by giving the administrator more control over early
2263 matches employing one of the short-circuit pattern match operators. Note
2264 that custom applications can trigger this same behavior by returning the
2265 special value AST_PBX_INCOMPLETE.
2269 * Directory now permits both first and last names to be matched at the same
2270 time. In addition, the number of digits to enter of the name can be set in
2271 the arguments to Directory; previously, you could enter only 3, regardless
2272 of how many names are in your company. For large companies, this should be
2274 * Voicemail now permits a mailbox setting to wrap around from first to last
2275 messages, if the "messagewrap" option is set to a true value.
2276 * Voicemail now permits an external script to be run, for password validation.
2277 The script should output "VALID" or "INVALID" on stdout, depending upon the
2278 wish to validate or invalidate the password given. Arguments are:
2279 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
2281 * Dial has a new option: F(context^extension^pri), which permits a callee to
2282 continue in the dialplan, at the specified label, if the caller hangs up.
2283 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
2284 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
2285 * The Jack application now has a c() option to supply a custom client name.
2286 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
2287 like the pre-existing whisper mode, except that the spy can also talk to the
2288 participant on the bridged channel as well.
2289 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
2290 to be spoken instead of the channel name or number. For more information on the
2291 use of this option, issue the command "core show application ChanSpy" from the
2293 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
2294 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
2295 words, if using the 'd' option, it is not possible to enter a number to append to
2296 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
2297 change to whisper mode, and pressing 6 will change to barge mode.
2298 * ExternalIVR now takes several options that affect the way it performs, as
2299 well as having several new commands. Please see the External IVR page on the Asterisk
2300 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
2301 * Added ability to communicate over a TCP socket instead of forking a child process for the
2302 ExternalIVR application.
2303 * ChanIsAvail has a new option, 'a', which will return all available channels instead
2304 of just the first one if you give the function more then one channel to check.
2305 * PrivacyManager now takes an option where you can specify a context where the
2306 given number will be matched. This way you have more control over who is allowed
2307 and it stops the people who blindly enter 10 digits.
2308 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
2309 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
2310 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
2311 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
2312 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
2313 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
2314 * The Dial() application no longer copies the language used by the caller to the callee's
2315 channel. If you desire for the caller's channel's language to be used for file playback
2316 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
2317 * SendImage() no longer hangs up the channel on error; instead, it sets the
2318 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
2319 'UNSUPPORTED'. This change makes SendImage() more consistent with other
2321 * Park has a new option, 's', which silences the announcement of the parking space number.
2322 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
2323 invalid input and will be assumed to mean that no timeout is desired.
2327 * Added DNS manager support to registrations for peers referencing peer entries.
2328 DNS manager runs in the background which allows DNS lookups to be run asynchronously
2329 as well as periodically updating the IP address. These properties allow for
2330 better performance as well as recovery in the event of an IP change.
2331 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
2332 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
2333 These changes also provide performance improvements for call setup and tear down.
2334 * Added ability to specify registration expiry time on a per registration basis in
2336 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
2338 * Added t38pt_usertpsource option. See sip.conf.sample for details.
2339 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
2340 * 'sip show peers' and 'sip show users' display their entries sorted in
2341 alphabetical order, as opposed to the order they were in, in the config
2343 * Videosupport now supports an additional option, "always", which always sets
2344 up video RTP ports, even on clients that don't support it. This helps with
2345 callfiles and certain transfers to ensure that if two video phones are
2346 connected, they will always share video feeds.
2350 * Existing DNS manager lookups extended to check for SRV records.
2351 * IAX2 encryption support has been improved to support periodic key rotation
2352 within a call for enhanced security. The option "keyrotate" has been
2353 provided to disable this functionality to preserve backwards compatibility
2354 with older versions of IAX2 that do not support key rotation.
2358 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
2359 data tree based on the given <path>.
2360 * New CLI command "data show providers" that will display all the registered
2362 * New CLI command, "config reload <file.conf>" which reloads any module that
2363 references that particular configuration file. Also added "config list"
2364 which shows which configuration files are in use.
2365 * New CLI commands, "pri show version" and "ss7 show version" that will
2366 display which version of libpri and libss7 are being used, respectively.
2367 A new API call was added so trunk will now have to be compiled against
2368 a versions of libpri and libss7 that have them or it will not know that
2369 these libraries exist.
2370 * The commands "core show globals", "core set global" and "core set chanvar" has
2371 been deprecated in favor of the more semanticly correct "dialplan show globals",
2372 "dialplan set chanvar" and "dialplan set global".
2373 * New CLI command "dialplan show chanvar" to list all variables associated
2374 with a given channel.
2378 * Addresses managed by DNS manager now can check to see if there is a DNS
2379 SRV record for a given domain and will use that hostname/port if present.
2381 AMI - The manager (TCP/TLS/HTTP)
2382 --------------------------------
2383 * The Status command now takes an optional list of variables to display
2384 along with channel status.
2385 * The QueueEntry event now also includes the channel's uniqueid
2389 * res_odbc no longer has a limit of 1023 total possible unshared connections,
2390 as some people were running into this limit. This limit has been increased
2395 * The TRANSFER queue log entry now includes the the caller's original
2396 position in the transferred-from queue.
2397 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
2398 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
2399 as well as an explanation about timeout options in general
2400 * Added a new option - C - for forcing the "answered elsewhere" flag on
2401 cancellation of calls in to members of the queue. This is to avoid the
2402 call to a member of a queue having the call listed as a "missed call".
2406 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
2407 adaptive capabilities. What this means in practical terms is that if your
2408 realtime table lacks critical fields, Asterisk will now emit warnings to
2409 that effect. Also, some of the realtime drivers have the ability (if
2410 configured) to automatically add those columns to the table with the
2411 correct type and length.
2415 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
2416 the 'setvar' option to cause a given audio file to be played upon completion
2417 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
2418 Skinny channels only.
2419 * You can now compile Asterisk against the Hoard Memory Allocator, see the
2420 Hoard page on the Asterisk wiki for more information:
2421 https://wiki.asterisk.org/wiki/x/pQBB
2422 * Config file variables may now be appended to, by using the '+=' append
2423 operator. This is most helpful when working with long SQL queries in
2424 func_odbc.conf, as the queries no longer need to be specified on a single
2426 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
2427 which will add a second to the billsec when the ending
2428 time is set, if the number in the microseconds field of the end time is
2429 greater than the number of microseconds in the answer time. This allows
2430 users to count the 'initiated' seconds in their billing records.
2432 ------------------------------------------------------------------------------
2433 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
2434 ------------------------------------------------------------------------------
2436 AMI - The manager (TCP/TLS/HTTP)
2437 --------------------------------
2438 * Manager has undergone a lot of changes, all of them documented
2439 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
2440 * Manager version has changed to 1.1
2441 * Added a new action 'CoreShowChannels' to list currently defined channels
2442 and some information about them.
2443 * Added a new action 'SIPshowregistry' to list SIP registrations.
2444 * Added TLS support for the manager interface and HTTP server
2445 * Added the URI redirect option for the built-in HTTP server
2446 * The output of CallerID in Manager events is now more consistent.
2447 CallerIDNum is used for number and CallerIDName for name.
2448 * Enable https support for builtin web server.
2449 See configs/http.conf.sample for details.
2450 * Added a new action, GetConfigJSON, which can return the contents of an
2451 Asterisk configuration file in JSON format. This is intended to help
2452 improve the performance of AJAX applications using the manager interface
2454 * SIP and IAX manager events now use "ChannelType" in all cases where we
2455 indicate channel driver. Previously, we used a mixture of "Channel"
2456 and "ChannelDriver" headers.
2457 * Added a "Bridge" action which allows you to bridge any two channels that
2458 are currently active on the system.
2459 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
2460 the voicemail users setup.
2461 * Added 'DBDel' and 'DBDelTree' manager commands.
2462 * cdr_manager now reports events via the "cdr" level, separating it from
2463 the very verbose "call" level.
2464 * Manager users are now stored in memory. If you change the manager account
2465 list (delete or add accounts) you need to reload manager.
2466 * Added Masquerade manager event for when a masquerade happens between
2468 * Added "manager reload" command for the CLI
2469 * Lots of commands that only provided information are now allowed under the
2470 Reporting privilege, instead of only under Call or System.
2471 * The IAX* commands now require either System or Reporting privilege, to
2472 mirror the privileges of the SIP* commands.
2473 * Added ability to retrieve list of categories in a config file.
2474 * Added ability to retrieve the content of a particular category.
2475 * Added ability to empty a context.
2476 * Created new action to create a new file.
2477 * Updated delete action to allow deletion by line number with respect to category.
2478 * Added new action insert to add new variable to category at specified line.
2479 * Updated action newcat to allow new category to be inserted in file above another
2481 * Added new event "JitterBufStats" in the IAX2 channel
2482 * Originate now requires the Originate privilege and, if you want to call out
2483 to a subshell, it requires the System privilege, as well. This was done to
2484 enhance manager security.
2485 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
2486 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
2487 or manager show command Atxfer from the CLI
2488 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
2489 details or manager show command IAXregistry from the CLI
2493 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
2494 state in the dialplan, as well as creating custom device states that are
2495 controllable from the dialplan.
2496 * Extend CALLERID() function with "pres" and "ton" parameters to
2497 fetch string representation of calling number presentation indicator
2498 and numeric representation of type of calling number value.
2499 * MailboxExists converted to dialplan function
2500 * A new option to Dial() for telling IP phones not to count the call
2501 as "missed" when dial times out and cancels.
2502 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
2503 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
2504 held for any given channel. Also, locks are automatically freed when a
2506 * Added HINT() dialplan function that allows retrieving hint information.
2507 Hints are mappings between extensions and devices for the sake of
2508 determining the state of an extension. This function can retrieve the list
2509 of devices or the name associated with a hint.
2510 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
2512 * Added SYSINFO() dialplan function which allows retrieval of system information
2513 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
2514 the existence of a dialplan target.
2515 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
2516 upper and lower case, respectively.
2517 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
2518 ID for the call (not the Asterisk call ID or unique ID), provided that the
2519 channel driver supports this. For SIP, you get the SIP call-ID for the
2520 bridged channel which you can store in the CDR with a custom field.
2524 * Added CLI permissions, config file: cli_permissions.conf
2525 default is to allow all commands for every local user/group.
2526 Also this new feature added three new CLI commands:
2527 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
2528 - cli reload permissions
2529 - cli show permissions
2530 * New CLI command "core show hint" (usage: core show hint <exten>)
2531 * New CLI command "core show settings"
2532 * Added 'core show channels count' CLI command.
2533 * Added the ability to set the core debug and verbose values on a per-file basis.
2534 * Added 'queue pause member' and 'queue unpause member' CLI commands
2535 * Ability to set process limits ("ulimit") without restarting Asterisk
2536 * Enhanced "agi debug" to print the channel name as a prefix to the debug
2537 output to make debugging on busy systems much easier.
2538 * New CLI commands "dialplan set extenpatternmatching true/false"
2539 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
2540 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
2541 listed in the startup_commands section of cli.conf will get executed.
2542 * Added a CLI command, "devstate change", which allows you to set custom device
2543 states from the func_devstate module that provides the DEVICE_STATE() function
2544 and handling of the "Custom:" devices.
2545 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
2546 sorted into the different possible callbacks, with the number of entries
2547 currently scheduled for each. Gives you a feel for how busy the sip channel
2549 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
2550 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
2551 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
2555 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
2556 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
2557 for a received call. If it is detected, the channel will jump to the
2558 'fax' extension in the dialplan.
2559 * The default SIP useragent= identifier now includes the Asterisk version
2560 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
2561 If set, and the incoming request carries authentication info,
2562 the username to match in the users list is taken from the Digest header
2563 rather than from the From: field. This feature is considered experimental.
2564 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
2565 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
2566 * The "localmask" setting was removed in version 1.2 and the reminder about it
2567 being removed is now also removed.
2568 * A new option "busylevel" for setting a level of calls where asterisk reports
2569 a device as busy, to separate it from call-limit. This value is also added
2570 to the SIP_PEER dialplan function.
2571 * A new realtime family called "sipregs" is now supported to store SIP registration
2572 data. If this family is defined, "sippeers" will be used for configuration and
2573 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
2574 registration data, as before.
2575 * The SIPPEER function have new options for port address, call and pickup groups
2576 * Added support for T.140 realtime text in SIP/RTP
2577 * The "checkmwi" option has been removed from sip.conf, as it is no longer
2578 required due to the restructuring of how MWI is handled. See the descriptions
2579 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
2580 for more information.
2581 * Added rtpdest option to CHANNEL() dialplan function.
2582 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
2583 * SIP now adds a header to the CANCEL if the call was answered by another phone
2584 in the same dial command, or if the new c option in dial() is used.
2585 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
2586 states it is not needed. For phones, however, that do require it the "registertrying" option
2587 has been added so it can be enabled.
2588 * A new option called "callcounter" (global/peer/user level) enables call counters needed
2589 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
2590 used to enable this functionality).
2591 * New settings for timer T1 and timer B on a global level or per device. This makes it
2592 possible to force timeout faster on non-responsive SIP servers. These settings are
2593 considered advanced, so don't use them unless you have a problem.
2594 * Added a dial string option to be able to set the To: header in an INVITE to any
2596 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
2597 the qualify frequency.
2598 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
2599 were not properly torn down due to network or endpoint failures during an established
2601 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
2602 and configs/sip.conf.sample for more information on how it is used.
2603 * Added a new configuration option "authfailureevents" that enables manager events when
2604 a peer can't authenticate properly.
2605 * Added DNS manager support to registrations for peers not referencing a peer entry.
2609 * Added the trunkmaxsize configuration option to chan_iax2.
2610 * Added the srvlookup option to iax.conf
2611 * Added support for OSP. The token is set and retrieved through the CHANNEL()
2614 XMPP Google Talk/Jingle changes
2615 -------------------------------
2616 * Added the bindaddr option to gtalk.conf.
2620 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
2621 * Proper codec support in chan_skinny.
2622 * Added settings for IP and Ethernet QoS requests
2626 * Added separate settings for media QoS in mgcp.conf
2628 Console Channel Driver changes
2629 ------------------------------
2630 * Added experimental support for video send & receive to chan_oss.
2631 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
2634 Phone channel changes (chan_phone)
2635 ----------------------------------
2636 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
2638 H.323 channel Changes
2639 ---------------------
2640 * H323 remote hold notification support added (by NOTIFY message
2641 and/or H.450 supplementary service)
2643 Local channel changes
2644 ---------------------
2645 * The device state functionality in the Local channel driver has been updated
2646 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
2647 to just UNKNOWN if the extension exists.
2648 * Added jitterbuffer support for chan_local. This allows you to use the
2649 generic jitterbuffer on incoming calls going to Asterisk applications.
2650 For example, this would allow you to use a jitterbuffer for an incoming
2651 SIP call to Voicemail by putting a Local channel in the middle. This
2652 feature is enabled by using the 'j' option in the Dial string to the Local
2653 channel in conjunction with the existing 'n' option for local channels.
2654 * A 'b' option has been added which causes chan_local to return the actual channel
2655 that is behind it when queried. This is useful for transfer scenarios as the
2656 actual channel will be transferred, not the Local channel.
2658 Agent channel changes
2659 ----------------------
2660 * The ackcall and endcall options are now supplemented with options acceptdtmf
2661 and enddtmf. These allow for the DTMF keypress to be configurable. The options
2662 default to their old hard-coded values ('#' and '*' respectively) so this should
2663 not break any existing agent installations.
2665 DAHDI channel driver (chan_dahdi) Changes
2666 ----------------------------------------
2667 * SS7 support (via libss7 library)
2668 * In India, some carriers transmit CID via dtmf. Some code has been added
2669 that will handle some situations. The cidstart=polarity_IN choice has been added for
2670 those carriers that transmit CID via dtmf after a polarity change.
2671 * CID matching information is now shown when doing 'dialplan show'.
2672 * Added dahdi show version CLI command.
2673 * Added setvar support to chan_dahdi.conf channel entries.
2674 * Added two new options: mwimonitor and mwimonitornotify. These options allow
2675 you to enable MWI monitoring on FXO lines. When the MWI state changes,
2676 the script specified in the mwimonitornotify option is executed. An internal
2677 event indicating the new state of the mailbox is also generated, so that
2678 the normal MWI facilities in Asterisk work as usual.
2679 * Added signalling type 'auto', which attempts to use the same signalling type
2680 for a channel as configured in DAHDI. This is primarily designed for analog
2681 ports, but will also work for digital ports that are configured for FXS or FXO
2682 signalling types. This mode is also the default now, so if your chan_dahdi.conf
2683 does not specify signalling for a channel (which is unlikely as the sample
2684 configuration file has always recommended specifying it for every channel) then
2685 the 'auto' mode will be used for that channel if possible.
2686 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
2687 state for a channel; also ensured that the DNDState Manager event is
2688 emitted no matter how the DND state is set or cleared.
2692 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
2693 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
2694 for details. This new channel driver allows you to use Nortel i2002,
2695 i2004, and i2050 phones with Asterisk.
2696 * Added a new channel driver, chan_console, which uses portaudio as a cross
2697 platform audio interface. It was written as a channel driver that would
2698 work with Mac CoreAudio, but portaudio supports a number of other audio
2699 interfaces, as well. Note that this channel driver requires v19 or higher
2700 of portaudio; older versions have a different API.
2704 * Added the ability to specify arguments to the Dial application when using
2705 the DUNDi switch in the dialplan.
2706 * Added the ability to set weights for responses dynamically. This can be
2707 done using a global variable or a dialplan function. Using the SHELL()
2708 function would allow you to have an external script set the weight for
2710 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
2711 functions will allow you to initiate a DUNDi query from the dialplan,
2712 find out how many results there are, and access each one.
2713 * Added the ability to specifiy a port for a dundi peer.
2717 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
2718 functions will allow you to initiate an ENUM lookup from the dialplan,
2719 and Asterisk will cache the results. ENUMRESULT can be used to access
2720 the results without doing multiple DNS queries.
2724 * Added the ability to customize which sound files are used for some of the
2725 prompts within the Voicemail application by changing them in voicemail.conf
2726 * Added the ability for the "voicemail show users" CLI command to show users
2727 configured by the dynamic realtime configuration method.
2728 * MWI (Message Waiting Indication) handling has been significantly
2729 restructured internally to Asterisk. It is now totally event based
2730 instead of polling based. The voicemail application will notify other
2731 modules that have subscribed to MWI events when something in the mailbox
2733 This also means that if any other entity outside of Asterisk is changing
2734 the contents of mailboxes, then the voicemail application still needs to
2735 poll for changes. Examples of situations that would require this option
2736 are web interfaces to voicemail or an email client in the case of using
2737 IMAP storage. So, two new options have been added to voicemail.conf
2738 to account for this: "pollmailboxes" and "pollfreq". See the sample
2739 configuration file for details.
2740 * Added "tw" language support
2741 * Added support for storage of greetings using an IMAP server
2742 * Added ability to customize forward, reverse, stop, and pause keys for message playback
2743 * SMDI is now enabled in voicemail using the smdienable option.
2744 * A "lockmode" option has been added to asterisk.conf to configure the file
2745 locking method used for voicemail, and potentially other things in the
2746 future. The default is the old behavior, lockfile. However, there is a
2747 new method, "flock", that uses a different method for situations where the
2748 lockfile will not work, such as on SMB/CIFS mounts.
2749 * Added the ability to backup deleted messages, to ease recovery in the case
2750 that a user accidentally deletes a message, and discovers that they need it.
2751 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
2752 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
2753 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
2754 voicemail boxes. The SMDI interface can also poll for MWI changes when some
2755 outside entity is modifying the state of the mailbox (such as IMAP storage or
2756 a web interface of some kind).
2757 * Added the support for marking messages as "urgent." There are two methods to accomplish
2758 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
2759 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
2760 the message as urgent after he has recorded a voicemail by following the voice instructions.
2761 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
2766 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
2767 used across multiple queues.
2768 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
2769 setqueueentryvar options for each queue, see queues.conf.sample for details.
2770 * Added keepstats option to queues.conf which will keep queue
2771 statistics during a reload.
2772 * setinterfacevar option in queues.conf also now sets a variable
2773 called MEMBERNAME which contains the member's name.
2774 * Added 'Strategy' field to manager event QueueParams which represents
2775 the queue strategy in use.
2776 * Added option to run macro when a queue member is connected to a caller,
2777 see queues.conf.sample for details.
2778 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
2779 does not count paused queue members as unavailable.
2780 * Added min-announce-frequency option to queues.conf which allows you to control the
2781 minimum amount of time between queue announcements for use when the caller's queue
2782 position changes frequently.
2783 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
2785 * Added ability for non-realtime queues to have realtime members
2786 * Added the "linear" strategy to queues.
2787 * Added the "wrandom" strategy to queues.
2788 * Added new channel variable QUEUE_MIN_PENALTY
2789 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
2790 rules in queuerules.conf. See configs/queuerules.conf.sample for details
2791 * Added a new parameter for member definition, called state_interface. This may be
2792 used so that a member may be called via one interface but have a different interface's
2793 device state reported.
2794 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
2795 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
2796 "manager show command QueueReset."
2797 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
2798 specified by the periodic-announce option, then one will be chosen randomly when it is time
2799 to play a periodic announcment
2800 * New configuration options: announce-position now takes two more values in addition to "yes" and
2801 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
2802 announce-position-limit. By setting announce-position to "limit" callers will only have their
2803 position announced if their position is less than what is specified by announce-position-limit.
2804 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
2805 will be told that their are more than announce-position-limit callers waiting.
2806 * Two new queue log events have been added. An ADDMEMBER event will be logged
2807 when a realtime queue member is added and a REMOVEMEMBER event will be logged
2808 when a realtime queue member is removed. Since there is no calling channel associated
2809 with these events, the string "REALTIME" is placed where the channel's unique id
2810 is typically placed.
2811 * The configuration method for the "joinempty" and "leavewhenempty" options has
2812 changed to a comma-separated list of methods of determining member availability
2813 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
2814 values are still accepted for backwards-compatibility, though.
2815 * The average talktime is now calculated on queues. This information is reported via the
2816 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
2817 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
2822 * The 'o' option to provide an optimization has been removed and its functionality
2823 has been enabled by default.
2824 * When a conference is created, the UNIQUEID of the channel that caused it to be
2825 created is stored. Then, every channel that joins the conference will have the
2826 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
2827 callers that come and go from long standing conferences.
2828 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
2829 except it does operations on a channel by name, instead of number in a conference.
2830 This is a very useful feature in combination with the 'X' option to ChanSpy.
2831 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
2833 * Added new RealTime functionality to provide support for scheduled conferencing.
2834 This includes optional messages to the caller if they attempt to join before
2835 the schedule start time, or to allow the caller to join the conference early.
2836 Also included is optional support for limiting the number of callers per
2837 RealTime conference.
2838 * Added the S() and L() options to the MeetMe application. These are pretty
2839 much identical to the S() and L() options to Dial(). They let you set
2840 timeouts for the conference, as well as have warning sounds played to
2841 let the caller know how much time is left, and when it is running out.
2842 * Added the ability to do "meetme concise" with the "meetme" CLI command.
2843 This extends the concise capabilities of this CLI command to include
2844 listing all conferences, instead of an addition to the other sub commands
2845 for the "meetme" command.
2846 * Added the ability to specify the music on hold class used to play into the
2847 conference when there is only one member and the M option is used.
2848 * Added MEETME_INFO dialplan function which provides a way to query
2849 various properties of a Meetme conference.
2850 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
2851 and *84: record in-conf
2853 Other Dialplan Application Changes
2854 ----------------------------------
2855 * Argument support for Gosub application
2856 * From the to-do lists: straighten out the app timeout args:
2857 Wait() app now really does 0.3 seconds- was truncating arg to an int.
2858 WaitExten() same as Wait().
2859 Congestion() - Now takes floating pt. argument.
2860 Busy() - now takes floating pt. argument.
2861 Read() - timeout now can be floating pt.
2862 WaitForRing() now takes floating pt timeout arg.
2863 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
2864 * Added 's' option to Page application.
2865 * Added an optional timeout argument to the Page application.
2866 * Added 'E', 'V', and 'P' commands to ExternalIVR.
2867 * Added 'o' and 'X' options to Chanspy.
2868 * Added a new dialplan application, Bridge, which allows you to bridge the
2869 calling channel to any other active channel on the system.
2870 * Added the ability to specify a music on hold class to play instead of ringing
2871 for the SLATrunk application.
2872 * The Read application no longer exits the dialplan on error. Instead, it sets
2873 READSTATUS to ERROR, which you can catch and handle separately.
2874 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
2875 of asking for verification of each name, one at a time.
2876 * Privacy() no longer uses privacy.conf, as all options are specifyable as
2877 direct options to the app.
2878 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
2880 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
2881 * The ChannelRedirect application no longer exits the dialplan if the given channel
2882 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
2883 or NOCHANNEL if the given channel was not found.
2884 * The silencethreshold setting that was previously configurable in multiple
2885 applications is now settable globally via dsp.conf.
2887 Music On Hold Changes
2888 ---------------------
2889 * A new option, "digit", has been added for music on hold classes in
2890 musiconhold.conf. If this is set for a music on hold class, a caller
2891 listening to music on hold can press this digit to switch to listening
2892 to this music on hold class.
2893 * Support for realtime music on hold has been added.
2894 * In conjunction with the realtime music on hold, a general section has
2895 been added to musiconhold.conf, its sole variable is cachertclasses. If this
2896 is set, then music on hold classes found in realtime will be cached in memory.
2900 * AEL upgraded to use the Gosub with Arguments instead
2901 of Macro application, to hopefully reduce the problems
2902 seen with the artificially low stack ceiling that
2903 Macro bumps into. Macros can only call other Macros
2904 to a depth of 7. Tests run using gosub, show depths
2905 limited only by virtual memory. A small test demonstrated
2906 recursive call depths of 100,000 without problems.
2907 -- in addition to this, all apps that allowed a macro
2908 to be called, as in Dial, queues, etc, are now allowing
2909 a gosub call in similar fashion.
2910 * AEL now generates LOCAL(argname) declarations when it
2911 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
2912 etc. That makes the arguments local in scope. The user
2913 can define their own local variables in macros, now,
2914 by saying "local myvar=someval;" or using Set() in this
2915 fashion: Set(LOCAL(myvar)=someval); ("local" is now
2917 * utils/conf2ael introduced. Will convert an extensions.conf
2918 file into extensions.ael. Very crude and unfinished, but
2919 will be improved as time goes by. Should be useful for a
2920 first pass at conversion.
2921 * aelparse will now read extensions.conf to see if a referenced
2922 macro or context is there before issueing a warning.
2923 * AEL parser sets a local channel variable ~~EXTEN~~, to
2924 preserve the value of ${EXTEN} thru switch statements.
2925 * New operator in $[...] expressions: the ~~ operator serves
2926 as a concatenation operator. AT THE MOMENT, it is really only
2927 necessary and useful in AEL, especially in if() expressions.
2928 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
2929 any enclosing double-quotes, and evaluate to the value of a
2930 concatenated with the value of b. For example if a is set to
2931 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
2932 evaluate to xyzabc .
2935 Call Features (res_features) Changes
2936 ------------------------------------
2937 * Added the parkedcalltransfers option to features.conf
2938 * Added parkedcallparking option to control one touch parking w/ parking
2940 * Added parkedcallhangup option to control disconnect feature w/ parking
2942 * Added parkedcallrecording option to control one-touch record w/ parking
2944 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
2945 parkedcalltransfers option support for multiple parking lots.
2946 * Added BRIDGE_FEATURES variable to set available features for a channel
2947 * The built-in method for doing attended transfers has been updated to
2948 include some new options that allow you to have the transferee sent
2949 back to the person that did the transfer if the transfer is not successful.
2950 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
2951 in features.conf.sample.
2952 * Added support for configuring named groups of custom call features in
2953 features.conf. This means that features can be written a single time, and
2954 then mapped into groups of features for different key mappings or easier
2956 * Updated the ParkedCall application to allow you to not specify a parking
2957 extension. If you don't specify a parking space to pick up, it will grab
2958 the first one available.
2959 * Added cli command 'features reload' to reload call features from features.conf
2960 * Moved into core asterisk binary.
2961 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
2962 * Added the ability for custom parking lots to be configured with their own
2963 parking extension with the parkext option.
2965 Language Support Changes
2966 ------------------------
2967 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
2968 * Added support for the Hungarian language for saying numbers, dates, and times.
2972 * Added SPEECH commands for speech recognition. A complete listing can be found
2974 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
2975 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
2976 does not behave as expected; the native command needs to be used, instead.
2977 * Added the ability to perform SRV lookups on fast AGI calls. To use this
2978 feature, simply use hagi: instead of agi: as the protocol portion
2979 of the URI parameter to the AGI function call in your dial plan. Also note
2980 that specifying a port number in the AGI URI will disable SRV lookups,
2981 even if you use the hagi: protocol.
2982 * No longer support MSG_OOB flag on HANGUP.
2986 * Added rotatestrategy option to logger.conf, along with two new options:
2987 "timestamp" which will use the time to name the logger files instead of
2988 sequence number; and "rotate", which rotates the names of the log files,
2989 similar to the way syslog rotates files.
2990 * Added exec_after_rotate option to logger.conf, which allows a system
2991 command to be run after rotation. This is primarily useful with
2992 rotatestrategy=rotate, to allow a limit on the number of log files kept
2993 and to ensure that the oldest log file gets deleted.
2994 * Added realtime support for the queue log
2998 * The cdr_manager module has a [mappings] feature, like cdr_custom,
2999 to add fields to the manager event from the CDR variables.
3000 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
3001 backend database CDR table. Specifically, additional, non-standard
3002 columns are supported, merely by setting the corresponding CDR variable in
3003 your dialplan. In addition, you may alias any column to another name (for
3004 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
3005 simply "alias src => ANI" in the configuration file). Records may be
3006 posted to more than one backend, simply by specifying multiple categories
3007 in the configuration file. And finally, you may filter which CDRs get
3008 posted to each backend, by specifying a filter (which the record must
3009 match) for the particular category. Filters are additive (meaning all
3010 rules must match to post that CDR).
3011 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
3012 module. Specifically, you may add additional columns into the table and
3013 they will be set, if you set the corresponding CDR variable name. Also,
3014 if you omit columns in your database table, they will be silently skipped
3015 (but a record will still be inserted, based on what columns remain). Note
3016 that the other two features from cdr_adaptive_odbc (alias and filter) are
3017 not currently supported.
3018 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
3019 has been disabled using the NoCDR application.
3021 Miscellaneous New Modules
3022 -------------------------
3023 * Added a new CDR module, cdr_sqlite3_custom.
3024 * Added a new realtime configuration module, res_config_sqlite
3025 * Added a new codec translation module, codec_resample, which re-samples
3026 signed linear audio between 8 kHz and 16 kHz to help support wideband
3028 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
3029 based on configuration templates that use Asterisk dialplan function and
3030 variable substitution. It should be possible to create phone profiles and
3031 templates that work for the majority of phones provisioned over http. It
3032 is currently only intended to provision a single user account per phone.
3033 An example profile and set of templates for Polycom phones is provided.
3034 NOTE: Polycom firmware is not included, but should be placed in
3035 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
3036 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
3037 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
3038 provided; there is a JACK() application, and a JACK_HOOK() function. Both
3039 interfaces create an input and output JACK port. The application makes
3040 these ports the endpoint of the call. The audio coming from the channel
3041 goes out the output port and whatever comes back in on the input port is
3042 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
3043 audiohook on the channel. This lets you run the audio coming from a
3044 channel through JACK, and whatever comes back in is what gets forwarded
3045 on as the channel's audio. This is very useful for building custom
3046 vocoders or doing recording or analysis of the channel's audio in another
3048 * Added a new module, res_config_curl, which permits using a HTTP POST url
3049 to retrieve, create, update, and delete realtime information from a remote
3050 web server. Note that this module requires func_curl.so to be loaded for
3051 backend functionality.
3052 * Added a new module, res_config_ldap, which permits the use of an LDAP
3053 server for realtime data access.
3054 * Added support for writing and running your dialplan in lua using the pbx_lua
3055 module. See configs/extensions.lua.sample for examples of how to do this.
3059 * Ability to use libcap to set high ToS bits when non-root
3060 on Linux. If configure is unable to find libcap then you
3061 can use --with-cap to specify the path.
3062 * Added maxfiles option to options section of asterisk.conf which allows you to specify
3063 what Asterisk should set as the maximum number of open files when it loads.
3064 * Added the jittertargetextra configuration option.
3065 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
3066 configuration files for the IP channel drivers. The new option is "cos".
3067 This information is also documented on the Asterisk wiki at
3068 https://wiki.asterisk.org/wiki/x/EYBG
3069 * When originating a call using AMI or pbx_spool that fails the reason for failure
3070 will now be available in the failed extension using the REASON dialplan variable.
3071 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
3072 It allows you to configure a prefix for auto-monitor recordings.
3073 * A new extension pattern matching algorithm, based on a trie, is introduced
3074 here, that could noticeably speed up mid-sized to large dialplans.
3075 It is NOT used by default, as duplicating the behaviour of the old pattern
3076 matcher is still under development. A config file option, in extensions.conf,
3077 in the [general] section, called "extenpatternmatchingnew", is by default
3078 set to false; setting that to true will force the use of the new algorithm.
3079 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
3080 be used to switch the algorithms at run time.
3081 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
3082 specifying which socket to use to connect to the running Asterisk daemon
3084 * Performance enhancements to the sched facility, which is used in
3085 the channel drivers, etc. Added hashtabs and doubly-linked lists
3086 to speed up deletion; start at the beginning or end of list to
3088 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
3089 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
3090 Added regression tests to the tests/ dir, also.
3091 * Added a refcount trace feature to astobj2 for those trying to balance
3092 object creation, deletion; work, play; space and time. See the
3093 notes in astobj2.h. Also, see utils/refcounter as well, as a
3094 quick way to find unbalanced refcounts in what could be a sea
3095 of objects that were balanced.
3096 * Added logging to 'make update' command. See update.log
3097 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
3098 do not come from the remote party.
3099 * Added the 'n' option to the SpeechBackground application to tell it to not
3100 answer the channel if it has not already been answered.
3101 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
3102 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
3104 * iLBC source code no longer included (see UPGRADE.txt for details)
3105 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
3106 deadlock is detected, a backtrace of the stack which led to the lock calls
3107 will be output to the CLI.
3108 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
3109 the "core show locks" CLI command will give lock information output as well
3110 as a backtrace of the stack which led to the lock calls.
3111 * users.conf now sports an optional alternateexts property, which permits
3112 allocation of additional extensions which will reach the specified user.
3113 * A new option for the configure script, --enable-internal-poll, has been added
3114 for use with systems which may have a buggy implementation of the poll system
3115 call. If you notice odd behavior such as the CLI being unresponsive on remote
3116 consoles, you may want to try using this option. This option is enabled by default
3117 on Darwin systems since it is known that the Darwin poll() implementation has
3121 --------------------
3122 * In addition to timing from DAHDI, there is a new timing module called
3123 res_timing_timerfd. In order to use this, you must be running Linux with
3124 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
3125 script will be able to tell if you have the requirements. From menuselect, select
3126 res_timing_timerfd from the Resource Modules menu.
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