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 ------------------------------------------------------------------------------
15 AMI (Asterisk Manager Interface)
17 * The SIPshowpeer action will now include a 'SubscribeContext' field for a peer
18 in its response if the peer has a subscribe context set.
20 * The SIPqualifypeer action now acknowledges the request once it has established
21 that the request is against a known peer. It also issues a new event,
22 'SIPqualifypeerdone', once the qualify action has been completed.
24 * The PlayDTMF action now supports an optional 'Duration' parameter. This
25 specifies the duration of the digit to be played, in milliseconds.
27 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
28 updates when changes occur instead of requiring the use of pollmailboxes.
30 * CLI Command 'Manager Show Commands' no longer truncates command names longer
31 than 15 characters and no longer shows authorization requirement for commands.
32 'Manager Show Command' now displays the privileges needed for using a given
33 manager command instead.
35 * Added new action "ControlPlayback". The ControlPlayback action allows an AMI
36 client to manipulate audio currently being played back on a channel. The
37 supported operations depend on the application being used to send audio to
38 the channel. When the audio playback was initiated using the ControlPlayback
39 application or CONTROL STREAM FILE AGI command, the audio can be paused,
40 stopped, restarted, reversed, or skipped forward. When initiated by other
41 mechanisms (such as the Playback application), the audio can be stopped,
42 reversed, or skipped forward.
44 * Channel related events now contain a snapshot of channel state, adding new
45 fields to many of these events.
47 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
48 in a future release. Please use the common 'Exten' field instead.
50 * The AMI event 'UserEvent' from app_userevent now contains the channel state
51 fields. The channel state fields will come before the body fields.
53 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
54 'UnParkedCall' have changed significantly in the new res_parking module.
55 First, channel snapshot data is included for both the parker and the parkee
56 in lieu of the "From" and "Channel" fields. They follow standard channel
57 snapshot format but each field is suffixed with 'Parker' or 'Parkee'
58 depending on which side it applies to. The 'Exten' field is replaced with
59 'ParkingSpace' since the registration of extensions as for parking spaces
60 is no longer mandatory.
62 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
63 fashion has changed the field names 'StartExten' and 'StopExten' to
64 'StartSpace' and 'StopSpace' respectively.
66 * The deprecated use of | (pipe) as a separator in the channelvars setting in
67 manager.conf has been removed.
69 * Channel Variables conveyed with a channel no longer contain the name of the
70 channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
71 ChanVariable: bar=baz. When multiple channels are present in a single AMI
72 event, the various ChanVariable fields will contain a suffix that specifies
73 which channel they correspond to.
75 * The AMI 'Status' response event to the AMI Status action replaces the
76 BridgedChannel and BridgedUniqueid headers with the BridgeID header to
77 indicate what bridge the channel is currently in.
81 * When a channel driver is configured to enable jiterbuffers, they are now
82 applied unconditionally when a channel joins a bridge. If a jitterbuffer
83 is already set for that channel when it enters, such as by the JITTERBUFFER
84 function, then the existing jitterbuffer will be used and the one set by
85 the channel driver will not be applied.
89 * The /b option is removed.
91 * chan_local moved into the system core and is no longer a loadable module.
95 * Added general support for busy detection.
97 * Added ECAM command support for Sony Ericsson phones.
101 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
102 using the 'supportpath' setting, either on a global basis or on a peer basis.
103 This setting enables Asterisk to route outgoing out-of-dialog requests via a
104 set of proxies by using a pre-loaded route-set defined by the Path headers in
105 the REGISTER request. See Realtime updates for more configuration information.
109 * The BRIDGE_FEATURES channel variable would previously only set features for
110 the calling party and would set this feature regardless of whether the
111 feature was in caps or in lowercase. Use of a caps feature for a letter
112 will now apply the feature to the calling party while use of a lowercase
113 letter will apply that feature to the called party.
115 * Add support for automixmonitor to the BRIDGE_FEATURES channel variable.
117 * Parking has been pulled from core and placed into a separate module called
118 res_parking. See Parking changes below for more details.
120 * You can now have the settings for a channel updated using the FEATURE()
121 and FEATUREMAP() functions inherited to child channels by setting
122 FEATURE(inherit)=yes.
126 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
127 to remove jitterbuffers previously set on a channel with JITTERBUFFER.
128 The value of this setting is ignored when disabled is used for the argument.
132 * When performing queue pause/unpause on an interface without specifying an
133 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
134 least one member of any queue exists for that interface.
136 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
137 for realtime queue log entries.
141 * Added the 'n' option to MeetMe to prevent application of the DENOISE function
142 to a channel joining a conference. Some channel drivers that vary the number
143 of audio samples in a voice frame will experience significant quality problems
144 if a denoiser is attached to the channel; this option gives them the ability
145 to remove the denoiser without having to unload func_speex.
149 * Parking is now implemented as a module instead of as core functionality.
150 The preferred way to configure parking is now through res_parking.conf while
151 configuration through features.conf is not currently supported.
153 * Parked calls are now placed in bridges. This is a largely architectural change,
154 but it could have some implications in allowing for new parked call retrieval
155 methods and the contents of parking lots will be visible though certain bridge
158 * The order of arguments for the new parking applications are different from the
159 old ones to be more intuitive. Timeout and return context/exten/priority are now
160 implemented as options. parking_lot_name is now the first parameter. See the
161 application documentation for Park, ParkedCall, and ParkAndAnnounce for more
162 in-depth information as well as syntax.
164 * Extensions are no longer automatically created in the dialplan to park calls,
165 pickup parked calls, etc by default.
167 * adsipark is no longer supported under the new parking model
169 * The PARKINGSLOT channel variable has been deprecated in favor of PARKING_SPACE
170 to match the naming scheme of the new system.
172 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
173 channel even when comebactoorigin=yes
175 * New CLI command 'parking show' allows you to inspect the currently in use
176 parking lots. 'parking show <parkinglot>' will also show the parked calls
177 in that specific parking lot.
179 * The CLI command 'parkedcalls' is now deprecated in favor of
180 'parking show <parkinglot>'.
182 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
183 can be used to get a list of parked calls only for a specific parking lot.
185 * The ParkAndAnnounce application is now provided through res_parking instead
186 of through the separate app_parkandannounce module.
188 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
189 by default. Instead, it will follow the timeout rules of the parking lot. The
190 old behavior can be reproduced by using the 'c' option.
194 * Add queue available hint. exten => 8501,hint,Queue:markq_avail
195 Note: the suffix '_avail' after the queuename.
196 Reports 'InUse' for no logged in agents or no free agents.
197 Reports 'Idle' when an agent is free.
201 * Redirecting reasons can now be set to arbitrary strings. This means
202 that the REDIRECTING dialplan function can be used to set the redirecting
203 reason to any string. It also allows for custom strings to be read as the
204 redirecting reason from SIP Diversion headers.
208 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
209 will store the path information for that peer when it registers. Realtime
210 tables can also use the 'supportpath' field to enable Path header support.
212 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
213 objectIdentifier. This maps to the supportpath option in sip.conf.
217 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
218 them, an Asterisk-specific version of pjproject needs to be installed.
219 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
223 * Device state for XMPP buddies is now available using the following format:
224 XMPP/<client name>/<buddy address>
225 If any resource is available the device state is considered to be not in use.
226 If no resources exist or all are unavailable the device state is considered
229 Security Events Framework
230 -------------------------
231 * Security Event timestamps now use ISO 8601 formatted date/time instead of the
232 "seconds-microseconds" format that it was using previously.
236 * All future modules which utilize Sorcery for object persistence must have a
237 column named "id" within their schema when using the Sorcery realtime module.
238 This column must be able to contain a string of up to 128 characters in length.
242 * UserEvent will now handle duplicate keys by overwriting the previous value
243 assigned to the key. UserEvent invocations will also be distributed to any
244 interested res_stasis applications.
246 ------------------------------------------------------------------------------
247 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
248 ------------------------------------------------------------------------------
254 * The Asterisk build system will now build and install a shared library
255 (libasteriskssl.so) used to wrap various initialization and shutdown functions
256 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
257 that Asterisk can ensure that these functions do *not* get called by any
258 modules that are loaded into Asterisk, since they should only be called once
259 in any single process. If desired, this feature can be disabled by supplying
260 the "--disable-asteriskssl" option to the configure script.
262 * A new make target, 'full', has been added to the Makefile. This performs
263 the same compilation actions as make all, but will also scan the entirety of
264 each source file for documentation. This option is needed to generate AMI
265 event documentation. Note that your system must have Python in order for
266 this make target to succeed.
268 * The optimization portion of the build system has been reworked to avoid
269 broken builds on certain architectures. All architecture-specific
270 optimization has been removed in favor of using -march=native to allow gcc
271 to detect the environment in which it is running when possible. This can
272 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
274 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
275 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
277 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
278 previously parsed the header file to obtain the version of Asterisk, you
279 will now have to go through Asterisk to get the version information.
287 * Added 'F()' option. Similar to the dial option, this can be supplied with
288 arguments indicating where the callee should go after the caller is hung up,
289 or without options specified, the priority after the Queue will be used.
294 * Added menu action admin_toggle_mute_participants. This will mute / unmute
295 all non-admin participants on a conference. The confbridge configuration
296 file also allows for the default sounds played to all conference users when
297 this occurs to be overriden using sound_participants_unmuted and
298 sound_participants_muted.
300 * Added menu action participant_count. This will playback the number of
301 current participants in a conference.
303 * Added announcement configuration option to user profile. If set the sound
304 file will be played to the user, and only the user, upon joining the
307 * Added record_file_append option that defaults to "yes", but if set to no
308 will create a new file between each start/stop recording.
313 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
314 channels respectively before the callee channels are called.
319 * Added support for IPv6.
321 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
322 external process will cause the current playlist to be cleared, including
323 stopping any audio file that is currently playing. This is useful when you
324 want to interrupt audio playback only when specific DTMF is entered by the
330 * A new option, 'I' has been added to app_followme. By setting this option,
331 Asterisk will not update the caller with connected line changes when they
332 occur. This is similar to app_dial and app_queue.
334 * The 'N' option is now ignored if the call is already answered.
336 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
337 and caller channels respectively before the callee channels are called.
339 * The winning FollowMe outgoing call is now put on hold if the caller put it on
345 * MixMonitor hooks now have IDs associated with them which can be used to
346 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
347 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
348 now accepts that ID as an argument.
350 * Added 'm' option, which stores a copy of the recording as a voicemail in the
356 * The connect action in app_mysql now allows you to specify a port number to
357 connect to. This is useful if you run a MySQL server on a non-standard
363 * Increased the default number of allowed destinations from 5 to 12.
368 * The app_page application now no longer depends on DAHDI or app_meetme. It
369 has been re-architected to use app_confbridge internally.
374 * Added queue options autopausebusy and autopauseunavail for automatically
375 pausing a queue member when their device reports busy or congestion.
377 * The 'ignorebusy' option for queue members has been deprecated in favor of
378 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
379 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
380 per interface basis. Individual ringinuse values can now be set in
381 queues.conf via an argument to member definitions. Lastly, the queue
382 'ringinuse' setting now only determines defaults for the per member
383 'ringinuse' setting and does not override per member settings like it does
386 * Added 'F()' option. Similar to the dial option, this can be supplied with
387 arguments indicating where the callee should go after the caller is hung up,
388 or without options specified, the priority after the Queue will be used.
390 * Added new option log_member_name_as_agent, which will cause the membername to
391 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
392 state_interface has been set.
394 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
396 * App_queue will now play periodic announcements for the caller that
397 holds the first position in the queue while waiting for answer.
401 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
402 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
403 changed arguments to SayUnixTime so that every option is truly optional even
404 when using multiple options (so that j option could be used without having to
405 manually specify timezone and format) There are other benefits, e.g., format
406 can now be used without specifying time zone as well.
411 * Addition of the VM_INFO function - see Function changes.
413 * The imapserver, imapport, and imapflags configuration options can now be
414 overriden on a user by user basis.
416 * When voicemail plays a message's envelope with saycid set to yes, when
417 reaching the caller id field it will play a recording of a file with the same
418 base name as the sender's callerid if there is a similarly named file in
419 <astspooldir>/recordings/callerids/
421 * Voicemails now contains a unique message identifier "msg_id", which is stored
422 in the message envelope with the sound files. IMAP backends will now store
423 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
424 backends will store the message identifier in a "msg_id" column. See
425 UPGRADE.txt for more information.
427 * Added VoiceMailPlayMsg application. This application will play a single
428 voicemail message from a mailbox. The result of the application, SUCCESS or
429 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
434 * Hangup handlers can be attached to channels using the CHANNEL() function.
435 Hangup handlers will run when the channel is hung up similar to the h
436 extension. The hangup_handler_push option will push a GoSub compatible
437 location in the dialplan onto the channel's hangup handler stack. The
438 hangup_handler_pop option will remove the last added location, and optionally
439 replace it with a new GoSub compatible location. The hangup_handler_wipe
440 option will remove all locations on the stack, and optionally add a new
443 * The expression parser now recognizes the ABS() absolute value function,
444 which will convert negative floating point values to positive values.
446 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
447 control of faxdetect.
449 * Addition of the VM_INFO function that can be used to retrieve voicemail
450 user information, such as the email address and full name.
451 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
454 * The REDIRECTING function now supports the redirecting original party id
457 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
458 lets you set some of the configuration options from the [general] section
459 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
460 the key sequence used to activate built-in features, such as blindxfer,
461 and automon. See the built-in documentation for details.
463 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
464 instead of simply the uri. This is the format that MessageSend() can use
465 in the from parameter for outgoing SIP messages.
467 * Added the PRESENCE_STATE function. This allows retrieving presence state
468 information from any presence state provider. It also allows setting
469 presence state information from a CustomPresence presence state provider.
470 See AMI/CLI changes for related commands.
472 * Added the AMI_CLIENT function to make manager account attributes available
473 to the dialplan. It currently supports returning the current number of
474 active sessions for a given account.
476 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
477 and the REDIRECTING functions.
485 * Added a manager event "LocalBridge" for local channel call bridges between
486 the two pseudo-channels created.
491 * Added dialtone_detect option for analog ports to disconnect incoming
492 calls when dialtone is detected.
494 * Added option colp_send to send ISDN connected line information. Allowed
495 settings are block, to not send any connected line information; connect, to
496 send connected line information on initial connect; and update, to send
497 information on any update during a call. Default is update.
499 * Add options namedcallgroup and namedpickupgroup to support installations
500 where a higher number of groups (>64) is required.
502 * Added support to use private party ID information with PRI calls.
507 * A new channel driver named chan_motif has been added which provides support for
508 Google Talk and Jingle in a single channel driver. This new channel driver includes
509 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
510 hold, unhold, and ringing notification. It is also compliant with the current Jingle
511 specification, current Google Jingle specification, and the original Google Talk
517 * Added NAT support for RTP. Setting in config is 'nat', which can be set
518 globally and overriden on a peer by peer basis.
520 * Direct media functionality has been added. Options in config are:
521 directmedia (directrtp) and directrtpsetup (earlydirect)
523 * ChannelUpdate events now contain a CallRef header.
528 * Asterisk will no longer substitute CID number for CID name in the display
529 name field if CID number exists without a CID name. This change improves
530 compatibility with certain device features such as Avaya IP500's directory
533 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
534 created using that setting to not be removed during SIP reload.
536 * Added settings recordonfeature and recordofffeature. When receiving an INFO
537 request with a "Record:" header, this will turn the requested feature on/off.
538 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
539 dynamic features must be enabled and configured properly on the requesting
540 channel for this to function properly.
542 * Add support to realtime for the 'callbackextension' option.
544 * When multiple peers exist with the same address, but differing
545 callbackextension options, incoming requests that are matched by address
546 will be matched to the peer with the matching callbackextension if it is
549 * Two new NAT options, auto_force_rport and auto_comedia, have been added
550 which set the force_rport and comedia options automatically if Asterisk
551 detects that an incoming SIP request crossed a NAT after being sent by
554 * The default global nat setting in sip.conf has been changed from force_rport
557 * NAT settings are now a combinable list of options. The equivalent of the
558 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
560 * Adds an option send_diversion which can be disabled to prevent
561 diversion headers from automatically being added to INVITE requests.
563 * Add support for lightweight NAT keepalive. If enabled a blank packet will
564 be sent to the remote host at a given interval to keep the NAT mapping open.
565 This can be enabled using the keepalive configuration option.
567 * Add option 'tonezone' to specify country code for indications. This option
568 can be set both globally and overridden for specific peers.
570 * The SIP Security Events Framework now supports IPv6.
572 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
573 between multiple user agents. When set, for directmedia reinvites,
574 Asterisk will not send an immediate reinvite on an incoming call leg. This
575 option is useful when peered with another SIP user agent that is known to
576 send immediate direct media reinvites upon call establishment.
578 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
581 * Add options subminexpiry and submaxexpiry to set limits of subscription
582 timer independently from registration timer settings. The setting of the
583 registration timer limits still is done by options minexpiry, maxexpiry
584 and defaultexpiry. For backwards compatibility the setting of minexpiry
585 and maxexpiry also is used to configure the subscription timer limits if
586 subminexpiry and submaxexpiry are not set in sip.conf.
588 * Set registration timer limits to default values when reloading sip
589 configuration and values are not set by configuration.
591 * Add options namedcallgroup and namedpickupgroup to support installations
592 where a higher number of groups (>64) is required.
594 * When a MESSAGE request is received, the address the request was received from
595 is now saved in the SIP_RECVADDR variable.
597 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
598 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
599 the ANI2/OLI information is set on the channel, which can be retrieved using
600 the CALLERID function.
602 * Peers can now be configured to support negotiation of ICE candidates using
603 the setting icesupport. See res_rtp_asterisk changes for more information.
605 * Added support for format attribute negotiation. See the Codecs changes for
608 * Extra headers specified with SIPAddHeader are sent with the REFER message
609 when using Transfer application. See refer_addheaders in sip.conf.sample.
611 * Added support to use private party ID information with calls.
613 * Adds an option discard_remote_hold_retrieval that when set stops telling
614 the peer to start music on hold.
619 * Added skinny version 17 protocol support.
624 * Added ability to use multiple lines for a single phone. This allows multiple
625 calls to occur on a single phone, using callwaiting and switching between calls.
627 * Added option 'sharpdial' allowing end dialing by pressing # key
629 * Added option 'interdigit_timer' to control phone dial timeout
631 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
633 * Added global 'debug' option, that enables debug in channel driver
635 * Added ability to translate on-screen menu in multiple languages. Tested on
636 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
637 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
640 * In addition to English added French and Russian languages for on-screen menus
642 * Reworked dialing number input: added dialing by timeout, immediate dial on
643 on dialplan compare, phone number length now not limited by screen size
645 * Added ability to pickup a call using features.conf defined value and
651 * Add options namedcallgroup and namedpickupgroup to support installations
652 where a higher number of groups (>64) is required.
654 * Added support to use private party ID information with calls.
659 * The minimum DTMF duration can now be configured in asterisk.conf
660 as "mindtmfduration". The default value is (as before) set to 80 ms.
661 (previously it was only available in source code)
663 * Named ACLs can now be specified in acl.conf and used in configurations that
664 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
665 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
666 working ACL. In addition, some CLI commands have been added to provide
667 show information and allow for module reloading - see CLI Changes.
669 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
670 items (separated by commas), and items in the rule can be negated by prefixing
671 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
672 longer necessray to control the order that the 'permit' and 'deny' columns are
673 returned from queries.
675 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
676 be used within the dynamic weight attribute when specifying a mapping.
678 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
679 header, instead of putting the user defined event name there. When enabled
680 the UserDefType header is added for user defined events. This feature is
681 enabled with the setting show_user_defined.
683 * Macro has been deprecated in favor of GoSub. For redirecting and connected
684 line purposes use the following variables instead of their macro equivalents:
685 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
686 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
687 cc_callback_macro in channel configurations.
689 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
692 * Call files now support the "early_media" option to connect with an outgoing
693 extension when early media is received.
695 * Added support to use private party ID information with calls.
700 * A new channel variable, AGIEXITONHANGUP, has been added which allows
701 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
702 AGI application would exit immediately after a channel hangup is detected.
704 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
705 are resolved and each address is attempted in turn until one succeeds or
709 AMI (Asterisk Manager Interface)
711 * The originate action now has an option "EarlyMedia" that enables the
712 call to bridge when we get early media in the call. Previously,
713 early media was disregarded always when originating calls using AMI.
715 * Added setvar= option to manager accounts (much like sip.conf)
717 * Originate now generates an error response if the extension given is not found
720 * MixMonitor will now show IDs associated with the mixmonitor upon creating
721 them if the i(variable) option is used. StopMixMonitor will accept
722 MixMonitorID as an option to close specific MixMonitors.
724 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
725 updated to include information about peers configured with
726 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
727 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
728 returned if auto_force_rport is not enabled.
730 * Added SIPpeerstatus manager command which will generate PeerStatus events
731 similar to the existing PeerStatus events found in chan_sip on demand.
733 * Hangup now can take a regular expression as the Channel option. If you want
734 to hangup multiple channels, use /regex/ as the Channel option. Existing
735 behavior to hanging up a single channel is unchanged, but if you pass a regex,
736 the manager will send you a list of channels back that were hung up.
738 * Support for IPv6 addresses has been added.
740 * AMI Events can now be documented in the Asterisk source. Note that AMI event
741 documentation is only generated when Asterisk is compiled using 'make full'.
742 See the CLI section for commands to display AMI event information.
744 * The AMI Hangup event now includes the AccountCode header so you can easily
745 correlate with AMI Newchannel events.
747 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
748 the StateInterface of the queue member.
750 * Added AMI event SessionTimeout in the Call category that is issued when a
751 call is terminated due to either RTP stream inactivity or SIP session timer
754 * CEL events can now contain a user defined header UserDefType. See core
755 changes for more information.
757 * OOH323 ChannelUpdate events now contain a CallRef header.
759 * Added PresenceState command. This command will report the presence state for
760 the given presence provider.
762 * Added Parkinglots command. This will list all parking lots as a series of
763 AMI Parkinglot events.
765 * Added MessageSend command. This behaves in the same manner as the
766 MessageSend application, and is a technolgoy agnostic mechanism to send out
767 of call text messages.
769 * Added "message" class authorization. This grants an account permission to
770 send out of call messages. Write-only.
775 * The "dialplan add include" command has been modified to create context a context
776 if one does not already exist. For instance, "dialplan add include foo into bar"
777 will create context "bar" if it does not already exist.
779 * A "dialplan remove context" command has been added to remove a context from
782 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
783 filenames of all running mixmonitors on a channel.
785 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
786 numeric instead of 0, 1, or 2.
788 * "stun show status" will show a table describing how the STUN client is
791 * "acl show [named acl]" will show information regarding a Named ACL. The
792 acl module can be reloaded with "reload acl".
794 * Added CLI command to display AMI event information - "manager show events",
795 which shows a list of all known and documented AMI events, and "manager show
796 event [event name]", which shows detail information about a specific AMI
799 * The result of the CLI command "queue show" now includes the state interface
800 information of the queue member.
802 * The command "core set verbose" will now set a separate level of logging for
803 each remote console without affecting any other console.
805 * Added command "cdr show pgsql status" to check connection status
807 * "sip show channel" will now display the complete route set.
809 * Added "presencestate list" command. This command will list all custom
810 presence states that have been set by using the PRESENCE_STATE dialplan
813 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
814 command. This changes a custom presence to a new state.
819 * Codec lists may now be modified by the '!' character, to allow succinct
820 specification of a list of codecs allowed and disallowed, without the
821 requirement to use two different keywords. For example, to specify all
822 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
824 * Add support for parsing SDP attributes, generating SDP attributes, and
825 passing it through. This support includes codecs such as H.263, H.264, SILK,
826 and CELT. You are able to set up a call and have attribute information pass.
827 This should help considerably with video calls.
829 * The iLBC codec can now use a system-provided iLBC library if one is installed,
830 just like the GSM codec.
834 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
835 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
839 * Asterisk version and build information is now logged at the beginning of a
842 * Threads belonging to a particular call are now linked with callids which get
843 added to any log messages produced by those threads. Log messages can now be
844 easily identified as involved with a certain call by looking at their call id.
845 Call ids may also be attached to log messages for just about any case where
846 it can be determined to be related to a particular call.
848 * Each logging destination and console now have an independent notion of the
849 current verbosity level. Logger.conf now allows an optional argument to
850 the 'verbose' specifier, indicating the level of verbosity sent to that
851 particular logging destination. Additionally, remote consoles now each
852 have their own verbosity level. The command 'core set verbose' will now set
853 a separate level for each remote console without affecting any other
859 * Added 'announcement' option which will play at the start of MOH and between
860 songs in modes of MOH that can detect transitions between songs (eg.
866 * New per parking lot options: comebackcontext and comebackdialtime. See
867 configs/features.conf.sample for more details.
869 * Channel variable PARKER is now set when comebacktoorigin is disabled in
872 * Channel variable PARKEDCALL is now set with the name of the parking lot
873 when a timeout occurs.
879 CDR Postgresql Driver
881 * Added command "cdr show pgsql status" to check connection status
884 CDR Adaptive ODBC Driver
886 * Added schema option for databases that support specifying a schema.
894 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
895 CALENDAR_WRITE has completed successfully.
900 * A new option, 'probation' has been added to rtp.conf
901 RTP in strictrtp mode can now require more than 1 packet to exit learning
902 mode with a new source (and by default requires 4). The probation option
903 allows the user to change the required number of packets in sequence to any
904 desired value. Use a value of 1 to essentially restore the old behavior.
905 Also, with strictrtp on, Asterisk will now drop all packets until learning
906 mode has successfully exited. These changes are based on how pjmedia handles
907 media sources and source changes.
909 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
910 enabled or disabled using the icesupport setting. A variety of other
911 settings have been introduced to configure STUN/TURN connections.
916 * A new module, res_corosync, has been introduced. This module uses the
917 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
918 of Asterisk servers to both Message Waiting Indication (MWI) and/or
919 Device State (presence) information. This module is very similar to, and
920 is a replacement for the res_ais module that was in previous releases of
926 * This module adds a cleaned up, drop-in replacement for res_jabber called
927 res_xmpp. This provides the same externally facing functionality but is
928 implemented differently internally. res_jabber has been deprecated in favor
929 of res_xmpp; please see the UPGRADE.txt file for more information.
934 * The safe_asterisk script has been updated to allow several of its parameters
935 to be set from environment variables. This also enables a custom run
936 directory of Asterisk to be specified, instead of defaulting to /tmp.
938 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
939 its value to determine the directory to assume is the top-level directory of
940 the source tree. If the variable is not set, it defaults to the current
941 behavior and uses the current working directory.
943 ------------------------------------------------------------------------------
944 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
945 ------------------------------------------------------------------------------
949 * Asterisk now has protocol independent support for processing text messages
950 outside of a call. Messages are routed through the Asterisk dialplan.
951 SIP MESSAGE and XMPP are currently supported. There are options in
952 jabber.conf and sip.conf to allow enabling these features.
953 -> jabber.conf: see the "sendtodialplan" and "context" options.
954 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
955 and "outofcall_message_context" options.
956 The MESSAGE() dialplan function and MessageSend() application have been
957 added to go along with this functionality. More detailed usage information
958 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
959 * If real-time text support (T.140) is negotiated, it will be preferred for
960 sending text via the SendText application. For example, via SIP, messages
961 that were once sent via the SIP MESSAGE request would be sent via RTP if
962 T.140 text is negotiated for a call.
966 * parkedmusicclass can now be set for non-default parking lots.
968 Asterisk Manager Interface
969 --------------------------
970 * PeerStatus now includes Address and Port.
971 * Added Hold events for when the remote party puts the call on and off hold
972 for chan_dahdi ISDN channels.
973 * Added new action MeetmeListRooms to list active conferences (shows same
974 data as "meetme list" at the CLI).
975 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
976 Description field that is set by 'description' in the channel configuration
978 * Added Uniqueid header to UserEvent.
979 * Added new action FilterAdd to control event filters for the current session.
980 This requires the system permission and uses the same filter syntax as
981 filters that can be defined in manager.conf
982 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
983 versions had some instances of the event converted, but others were left
984 as-is. All Unlink events should now be converted to Bridge events. The AMI
985 protocol version number was incremented to 1.2 as a result of this change.
988 --------------------------
989 * The HTTP Server can bind to IPv6 addresses.
992 --------------------------
993 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
994 with busydetect. usage example: busypattern=200,200,200,600
997 --------------------------
998 * New 'gtalk show settings' command showing the current settings loaded from
1000 * The 'logger reload' command now supports an optional argument, specifying an
1001 alternate configuration file to use.
1002 * 'dialplan add extension' command will now automatically create a context if
1003 the specified context does not exist with a message indicated it did so.
1004 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
1005 Description field which can be populated with 'description' in the channel
1006 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
1009 --------------------------
1010 * The filter option in cdr_adaptive_odbc now supports negating the argument,
1011 thus allowing records which do NOT match the specified filter.
1012 * Added ability to log CONGESTION calls to CDR
1015 --------------------------
1016 * Ability to define custom SILK formats in codecs.conf.
1017 * Addition of speex32 audio format with translation.
1018 * CELT codec pass-through support and ability to define
1019 custom CELT formats in codecs.conf.
1020 * Ability to read raw signed linear files with sample rates
1021 ranging from 8khz - 192khz. The new file extensions introduced
1022 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
1023 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
1024 Skinny, H.323, etc) can still only support the following codecs:
1025 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
1026 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
1027 Video: h261, h263, h263p, h264, mpeg4
1032 --------------------------
1033 * New highly optimized and customizable ConfBridge application capable of
1034 mixing audio at sample rates ranging from 8khz-96khz.
1035 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
1036 and bridge profiles on a channel.
1037 * CONFBRIDGE_INFO dialplan function capable of retrieving information
1038 about a conference such as locked status and number of parties, admins,
1040 * Addition of video_mode option in confbridge.conf for adding video support
1041 into a bridge profile.
1042 * Addition of the follow_talker video_mode in confbridge.conf. This video
1043 mode dynamically switches the video feed to always display the loudest talker
1044 supplying video in the conference.
1048 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
1049 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
1050 variables from asterisk.conf.
1054 * Addition of the JITTERBUFFER dialplan function. This function allows
1055 for jitterbuffering to occur on the read side of a channel. By using
1056 this function conference applications such as ConfBridge and MeetMe can
1057 have the rx streams jitterbuffered before conference mixing occurs.
1058 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
1060 * Added STRREPLACE function. This function let's the user search a variable
1061 for a given string to replace with another string as many times as the
1062 user specifies or just throughout the whole string.
1063 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
1064 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
1065 * Added extensions to chan_ooh323 in function CHANNEL()
1067 libpri channel driver (chan_dahdi) DAHDI changes
1068 --------------------------
1069 * Added moh_signaling option to specify what to do when the channel's bridged
1070 peer puts the ISDN channel on hold.
1071 * Added display_send and display_receive options to control how the display ie
1072 is handled. To send display text from the dialplan use the SendText()
1073 application when the option is enabled.
1074 * Added mcid_send option to allow sending a MCID request on a span.
1077 --------------------------
1078 * Added setvar option to calendar.conf to allow setting channel variables on
1079 notification channels.
1080 * Added "calendar show types" CLI command to list registered calendar
1084 --------------------------
1085 * Added two new options, r and t with file name arguments to record
1086 single direction (unmixed) audio recording separate from the bidirectional
1087 (mixed) recording. The mixed file name argument is optional now as long
1088 as at least one recording option is used.
1091 --------------------------
1092 * Added a new option, l, which will disable local call optimization for
1093 channels involved with the FollowMe thread. Use this option to improve
1094 compatability for a FollowMe call with certain dialplan apps, options, and
1098 --------------------------
1099 * Added option "k" that will automatically close the conference when there's
1100 only one person left when a user exits the conference.
1103 --------------------------
1104 * cel_pgsql now supports the 'extra' column for data added using the
1105 CELGenUserEvent() application.
1108 --------------------------
1109 * Support for defining hints has been added to pbx_lua. See the 'hints' table
1110 in the sample extensions.lua file for syntax details.
1111 * Applications that perform jumps in the dialplan such as Goto will now
1112 execute properly. When pbx_lua detects that the context, extension, or
1113 priority we are executing on has changed it will immediately return control
1114 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
1115 the priority after the currently executing priority.
1116 * An autoservice is now started by default for pbx_lua channels. It can be
1117 stopped and restarted using the autoservice_stop() and autoservice_start()
1121 --------------------------
1122 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
1123 into a FAXStatus event with an 'Operation' header that will be either
1124 'send', 'receive', and 'gateway'.
1125 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
1126 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
1127 feature will handle converting a fax call between an audio T.30 fax terminal
1128 and an IFP T.38 fax terminal.
1132 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
1133 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
1134 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
1138 * Added general option negative_penalty_invalid default off. when set
1139 members are seen as invalid/logged out when there penalty is negative.
1140 for realtime members when set remove from queue will set penalty to -1.
1141 * Added queue option autopausedelay when autopause is enabled it will be
1142 delayed for this number of seconds since last successful call if there
1143 was no prior call the agent will be autopaused immediately.
1144 * Added member option ignorebusy this when set and ringinuse is not
1145 will allow per member control of multiple calls as ringinuse does for
1150 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
1152 * Added 'k' option to MeetMe to automatically kill the conference when there's only
1153 one participant left (much like a normal call bridge)
1154 * Added extra argument to Originate to set timeout.
1158 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
1159 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
1160 utility in the UTILS section of menuselect. If an existing astdb is found and no
1161 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
1162 convert an existing astdb to the SQLite3 version automatically at runtime.
1166 * Modules marked as deprecated are no longer marked as building by default. Enabling
1167 these modules is still available via menuselect.
1171 * authdebug is now disabled by default. To enable this functionaility again
1172 set authdebug = yes in iax.conf.
1176 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
1177 releases it was disabled.
1181 * The PBX core previously made a call with a non-existing extension test for
1182 extension s@default and jump there if the extension existed.
1183 This was a bad default behaviour and violated the principle of least surprise.
1184 It has therefore been changed in this release. It may affect some
1185 applications and configurations that rely on this behaviour. Most channel
1186 drivers have avoided this for many releases by testing whether the extension
1187 called exists before starting the PBX and generating a local error.
1188 This behaviour still exists and works as before.
1190 Extension "s" is used when no extension is given in a channel driver,
1191 like immediate answer in DAHDI or calling to a domain with no user part
1194 ------------------------------------------------------------------------------
1195 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
1196 ------------------------------------------------------------------------------
1200 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
1201 now defaults to force_rport. It is very important that phones requiring nat=no be
1202 specifically set as such instead of relying on the default setting. If at all
1203 possible, all devices should have nat settings configured in the general section as
1204 opposed to configuring nat per-device.
1205 * Added preferred_codec_only option in sip.conf. This feature limits the joint
1206 codecs sent in response to an INVITE to the single most preferred codec.
1207 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
1208 to be used for the outgoing call. It must be one of the codecs configured
1210 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
1211 to be used for holding a private key. If tlsprivatekey is not specified,
1212 tlscertfile is searched for both public and private key.
1213 * Added tlsclientmethod option to sip.conf. This allows the protocol for
1214 outbound client connections to be specified.
1215 * The sendrpid parameter has been expanded to include the options
1216 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
1217 header to be sent (equivalent to setting sendrpid=yes) and setting
1218 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
1219 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
1220 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
1221 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
1222 will accept the SDP even if the SDP version number is not properly incremented,
1223 but will generate a warning in the log indicating that the SIP peer that sent
1224 the SDP should have the 'ignoresdpversion' option set.
1225 * The 'nat' option has now been been changed to have yes, no, force_rport, and
1226 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
1227 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
1228 remote side requests it and disables symmetric RTP support. Setting it to
1229 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
1230 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
1231 and enables symmetric RTP support.
1232 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
1233 response. This permits the master channel to know how each channel dialled
1234 in a multi-channel setup resolved in an individual way. This carries a
1235 performance penalty and can be disabled in sip.conf using the
1236 'storesipcause' option.
1237 * Added 'externtcpport' and 'externtlsport' options to allow custom port
1238 configuration for the externip and externhost options when tcp or tls is used.
1239 * Added support for message body (stored in content variable) to SIP NOTIFY message
1240 accessible via AMI and CLI.
1241 * Added 'media_address' configuration option which can be used to explicitly specify
1242 the IP address to use in the SDP for media (audio, video, and text) streams.
1243 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
1244 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
1246 * Added 'use_q850_reason' configuration option for generating and parsing
1247 if available Reason: Q.850;cause=<cause code> header. It is implemented
1248 in some gateways for better passing PRI/SS7 cause codes via SIP.
1249 * When dialing SIP peers, a new component may be added to the end of the dialstring
1250 to indicate that a specific remote IP address or host should be used when dialing
1251 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
1252 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
1253 ability to selectively force bridged channels to also be encrypted is also
1254 implemented. Branching in the dialplan can be done based on whether or not
1255 a channel has secure media and/or signaling.
1256 * Added directmediapermit/directmediadeny to limit which peers can send direct media
1258 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
1259 Charge messages to snom phones.
1260 * Added support for G.719 media streams.
1261 * Added support for 16khz signed linear media streams.
1262 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
1263 RTP has been outfitted with the same abilities.
1264 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
1265 available in device configurations as well as in the dial plan.
1266 * Addition of the 'subscribe_network_change' option for turning on and off
1267 res_stun_monitor module support in chan_sip.
1268 * Addition of the 'auth_options_requests' option for turning on and off
1269 authentication for OPTIONS requests in chan_sip.
1273 * Add #tryinclude statement for config files. This provides the same
1274 functionality as the #include statement however an asterisk module will
1275 still load if the filename does not exist. Using the #include statement
1276 Asterisk will not allow the module to load.
1280 * Added rtsavesysname option into iax.conf to allow the systname to be saved
1281 on realtime updates.
1282 * Added the ability for chan_iax2 to inform the dialplan whether or not
1283 encryption is being used. This interoperates with the SIP SRTP implementation
1284 so that a secure SIP call can be bridged to a secure IAX call when the
1285 dialplan requires bridged channels to be "secure".
1286 * Addition of the 'subscribe_network_change' option for turning on and off
1287 res_stun_monitor module support in chan_iax.
1292 * Added ability to preset channel variables on indicated lines with the setvar
1293 configuration option. Also, clearvars=all resets the list of variables back
1295 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
1296 See configs/res_pktccops.conf for more information.
1298 XMPP Google Talk/Jingle changes
1299 -------------------------------
1300 * Added the externip option to gtalk.conf.
1301 * Added the stunaddr option to gtalk.conf which allows for the automatic
1302 retrieval of the external ip from a stun server.
1306 * Added 'p' option to PickupChan() to allow for picking up channel by the first
1307 match to a partial channel name.
1308 * Added .m3u support for Mp3Player application.
1309 * Added progress option to the app_dial D() option. When progress DTMF is
1310 present, those values are sent immediately upon receiving a PROGRESS message
1311 regardless if the call has been answered or not.
1312 * Added functionality to the app_dial F() option to continue with execution
1313 at the current location when no parameters are provided.
1314 * Added the 'a' option to app_dial to answer the calling channel before any
1315 announcements or macros are executed.
1316 * Modified app_dial to set answertime when the called channel answers even if
1317 the called channel hangs up during playback of an announcement.
1318 * Modified app_dial 'r' option to support an additional parameter to play an
1319 indication tone from indications.conf
1320 * Added c() option to app_chanspy. This option allows custom DTMF to be set
1321 to cycle through the next available channel. By default this is still '*'.
1322 * Added x() option to app_chanspy. This option allows DTMF to be set to
1323 exit the application.
1324 * The Voicemail application has been improved to automatically ignore messages
1325 that only contain silence.
1326 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
1327 associated mailbox(es) to be greetings-only.
1328 * The ChanSpy application now has the 'S' option, which makes the application
1329 automatically exit once it hits a point where no more channels are available
1331 * The ChanSpy application also now has the 'E' option, which spies on a single
1332 channel and exits when that channel hangs up.
1333 * The MeetMe application now turns on the DENOISE() function by default, for
1334 each participant. In our tests, this has significantly decreased background
1335 noise (especially noisy data centers).
1336 * Voicemail now permits storage of secrets in a separate file, located in the
1337 spool directory of each individual user. The control for this is located in
1338 the "passwordlocation" option in voicemail.conf. Please see the sample
1339 configuration for more information.
1340 * The ChanIsAvail application now exposes the returned cause code using a separate
1341 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
1342 * Added 'd' option to app_followme. This option disables the "Please hold"
1344 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
1345 received will terminate recording.
1346 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
1347 Previously the folder could only be set per context, but has now been extended
1348 using the imapfolder option.
1349 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
1350 * Voicemail now allows the pager date format to be specified separately from the
1352 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
1353 to allow joining, leaving, and sending text to group chats.
1354 * MeetMe has a new option 'G' to play an announcement before joining a conference.
1355 * Page has a new option 'A(x)' which will playback an announcement simultaneously
1356 to all paged phones (and optionally excluding the caller's one using the new
1357 option 'n') before the call is bridged.
1358 * The 'f' option to Dial has been augmented to take an optional argument. If no
1359 argument is provided, the 'f' option works as it always has. If an argument is
1360 provided, then the connected party information of all outgoing channels created
1361 during the Dial will be set to the argument passed to the 'f' option.
1362 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
1364 * The OSP lookup application adds in/outbound network ID, optional security,
1365 number portability, QoS reporting, destination IP port, custom info and service
1367 * Added new application VMSayName that will play the recorded name of the voicemail
1368 user if it exists, otherwise will play the mailbox number.
1369 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
1370 retrieve state for a particular bridge, where <name> is the conference name
1371 * app_directory now allows exiting at any time using the operator or pound key.
1372 * Voicemail now supports setting a locale per-mailbox.
1373 * Two new applications are provided for declining counting phrases in multiple
1374 languages. See the application notes for SayCountedNoun and SayCountedAdj for
1376 * Voicemail now runs the externnotify script when pollmailboxes is activated and
1378 * Voicemail now includes rdnis within msgXXXX.txt file.
1379 * ExternalIVR now supports IPv6 addresses.
1380 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
1381 at https://wiki.asterisk.org/wiki/x/oQBB
1382 * ParkedCall and Park can now specify the parking lot to use.
1386 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
1387 over SRV records associated with a specific service. From the CLI, type
1388 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
1389 details on how these may be used.
1390 * PITCH_SHIFT dialplan function added. This function can be used to modify the
1391 pitch of a channel's tx and rx audio streams.
1392 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
1393 setting various connected line and redirecting party information.
1394 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
1395 support ISDN subaddressing.
1396 * The CHANNEL() function now supports the "name" and "checkhangup" options.
1397 * For DAHDI channels, the CHANNEL() dialplan function now allows
1398 the dialplan to request changes in the configuration of the active
1399 echo canceller on the channel (if any), for the current call only.
1402 exten => s,n,Set(CHANNEL(echocan_mode)=off)
1404 The possible values are:
1406 on - normal mode (the echo canceller is actually reinitialized)
1408 fax - FAX/data mode (NLP disabled if possible, otherwise completely
1410 voice - voice mode (returns from FAX mode, reverting the changes that
1411 were made when FAX mode was requested)
1412 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
1413 and setting variables on the channel which created the current channel.
1414 Administrators should take care to avoid naming conflicts, when multiple
1415 channels are dialled at once, especially when used with the Local channel
1416 construct (which all could set variables on the master channel). Usage
1417 of the HASH() dialplan function, with the key set to the name of the slave
1418 channel, is one approach that will avoid conflicts.
1419 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
1421 * func_odbc now allows multiple row results to be retrieved without using
1422 mode=multirow. If rowlimit is set, then additional rows may be retrieved
1423 from the same query by using the name of the function which retrieved the
1424 first row as an argument to ODBC_FETCH().
1425 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
1426 dialplan. This function returns the content of the received message.
1427 * Added REPLACE, which searches a given variable name for a set of characters,
1428 then either replaces them with a single character or deletes them.
1429 * Added PASSTHRU, which literally passes the same argument back as its return
1430 value. The intent is to be able to use a literal string argument to
1431 functions that currently require a variable name as an argument.
1432 * HASH-associated variables now can be inherited across channel creation, by
1433 prefixing the name of the hash at assignment with the appropriate number of
1434 underscores, just like variables.
1435 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
1436 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
1437 whether or not channels that are bridged to the current channel will be
1438 required to have secure signaling and/or media.
1439 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
1440 the current channel has secure signaling and/or media.
1441 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
1442 "no_media_path" option.
1443 Returns "0" if there is a B channel associated with the call.
1444 Returns "1" if no B channel is associated with the call. The call is either
1445 on hold or is a call waiting call.
1446 * Added option to dialplan function CDR(), the 'f' option
1447 allows for high resolution times for billsec and duration fields.
1448 * FILE() now supports line-mode and writing.
1449 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
1450 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
1454 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
1455 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
1456 and is set when a dynamic feature is triggered.
1457 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
1458 to dynamically create a new parking lot matching the value this varible is
1460 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
1461 features.conf that should be the base for dynamic parkinglots.
1462 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
1463 parkinglot should have.
1464 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
1465 parkinglot should have.
1466 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
1471 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
1472 timeout has expired.
1473 * Added 'R' option to app_queue. This option stops moh and indicates ringing
1474 to the caller when an Agent's phone is ringing. This can be used to indicate
1475 to the caller that their call is about to be picked up, which is nice when
1476 one has been on hold for an extened period of time.
1477 * A new config option, penaltymemberslimit, has been added to queues.conf.
1478 When set this option will disregard penalty settings when a queue has too
1480 * A new option, 'I' has been added to both app_queue and app_dial.
1481 By setting this option, Asterisk will not update the caller with
1482 connected line changes or redirecting party changes when they occur.
1483 * A 'relative-periodic-announce' option has been added to queues.conf. When
1484 enabled, this option will cause periodic announce times to be calculated
1485 from the end of announcements rather than from the beginning.
1486 * The autopause option in queues.conf can be passed a new value, "all." The
1487 result is that if a member becomes auto-paused, he will be paused in all
1488 queues for which he is a member, not just the queue that failed to reach
1490 * Added dialplan function QUEUE_EXISTS to check if a queue exists
1491 * The queue logger now allows events to optionally propagate to a file,
1492 even when realtime logging is turned on. Additionally, realtime logging
1493 supports sending the event arguments to 5 individual fields, although it
1494 will fallback to the previous data definition, if the new table layout is
1497 mISDN channel driver (chan_misdn) changes
1498 ----------------------------------------
1499 * Added display_connected parameter to misdn.conf to put a display string
1500 in the CONNECT message containing the connected name and/or number if
1501 the presentation setting permits it.
1502 * Added display_setup parameter to misdn.conf to put a display string
1503 in the SETUP message containing the caller name and/or number if the
1504 presentation setting permits it.
1505 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
1506 indicate the dialplan settings are to be obtained from the asterisk
1508 * Made misdn.conf parameter callerid accept the "name" <number> format
1509 used by the rest of the system.
1510 * Made use the nationalprefix and internationalprefix misdn.conf
1511 parameters to prefix any received number from the ISDN link if that
1512 number has the corresponding Type-Of-Number. NOTE: This includes
1513 comparing the incoming call's dialed number against the MSN list.
1514 * Added the following new parameters: unknownprefix, netspecificprefix,
1515 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
1516 received number from the ISDN link if that number has the corresponding
1518 * Added new dialplan application misdn_command which permits controlling
1519 the CCBS/CCNR functionality.
1520 * Added new dialplan function mISDN_CC which permits retrieval of various
1521 values from an active call completion record.
1522 * For PTP, you should manually send the COLR of the redirected-to party
1523 for an incomming redirected call if the incoming call could experience
1524 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
1525 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
1526 if the REDIRECTING(from-num) is not empty.
1527 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
1528 option on all of the REDIRECTING statements before dialing the
1529 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
1530 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
1531 redirecting-to presentation (COLR) when it becomes available.
1532 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
1535 thirdparty mISDN enhancements
1536 -----------------------------
1537 mISDN has been modified by Digium, Inc. to greatly expand facility message
1539 * Enhanced COLP support for call diversion and transfer.
1540 * CCBS/CCNR support.
1542 The latest modified mISDN v1.1.x based version is available at:
1543 http://svn.digium.com/svn/thirdparty/mISDN/trunk
1544 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
1546 Tagged versions of the modified mISDN code are available under:
1547 http://svn.digium.com/svn/thirdparty/mISDN/tags
1548 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
1550 libpri channel driver (chan_dahdi) DAHDI changes
1551 -------------------------------------------
1552 * The channel variable PRIREDIRECTREASON is now just a status variable
1553 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
1554 to read and alter the reason.
1555 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
1556 redirected-to party for an incomming redirected call if the incoming call
1557 could experience further redirects. Just set the
1558 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
1559 to the COLR. A call has been redirected if the REDIRECTING(count) is not
1561 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
1562 use the inhibit(i) option on all of the REDIRECTING statements before
1563 dialing the redirected-to party. You still have to set the
1564 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
1565 will update the redirecting-to presentation (COLR) when it becomes available.
1566 * Added the ability to ignore calls that are not in a Multiple Subscriber
1567 Number (MSN) list for PTMP CPE interfaces.
1568 * Added dynamic range compression support for dahdi channels. It is
1569 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
1570 * Added support for ISDN calling and called subaddress with partial support
1571 for connected line subaddress.
1572 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
1573 * Added handling of received HOLD/RETRIEVE messages and the optional ability
1574 to transfer a held call on disconnect similar to an analog phone.
1575 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
1576 Will reroute/deflect an outgoing call when receive the message.
1577 Can use the DAHDISendCallreroutingFacility to send the message for the
1579 * Added standard location to add options to chan_dahdi dialing:
1580 Dial(DAHDI/g1[/extension[/options]])
1583 R Reverse charging indication
1584 * Added Reverse Charging Indication (Collect calls) send/receive option.
1585 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
1586 Dial(DAHDI/g1/extension/R)
1587 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
1588 (requires latest LibPRI)
1589 * Added ability to send/receive keypad digits in the SETUP message.
1590 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
1591 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
1592 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
1593 (requires latest LibPRI)
1594 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
1595 to eliminate tromboned calls. A tromboned call goes out an interface and comes
1596 back into the same interface. Tromboned calls happen because of call routing,
1597 call deflection, call forwarding, and call transfer.
1598 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
1599 * Added the ability to support call waiting calls. (The SETUP has no B channel
1601 * Added Malicious Call ID (MCID) event to the AMI call event class.
1602 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
1604 Asterisk Manager Interface
1605 --------------------------
1606 * The Hangup action now accepts a Cause header which may be used to
1607 set the channel's hangup cause.
1608 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
1609 to specify a separate .pem file to hold a private key. By default sslcert
1610 is used to hold both the public and private key.
1611 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
1612 for options containing the 'tls' prefix. For example, 'sslenable' is now
1613 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
1614 across all .conf files. All affected sample.conf files have been modified to
1615 reflect this change. Previous options such as 'sslenable' still work,
1616 but options with the 'tls' prefix are preferred.
1617 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
1618 in a channel. (res_mutestream.so)
1619 * The configuration file manager.conf now supports a channelvars option, which
1620 specifies a list of channel variables to include in each channel-oriented
1622 * The redirect command now has new parameters ExtraContext, ExtraExtension,
1623 and ExtraPriority to allow redirecting the second channel to a different
1624 location than the first.
1625 * Added new event "JabberStatus" in the Jabber module to monitor buddies
1627 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
1628 in a MixMonitor recording.
1629 * The 'iax2 show peers' output is now similar to the expected output of
1631 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
1633 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
1634 AOC-E messages on a channel.
1635 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
1636 conform more closely to similar events.
1637 * Added a new eventfilter option per user to allow whitelisting and blacklisting
1639 * Added optional parkinglot variable for park command.
1640 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
1641 if CallerIDNum and CallerIDName headers are also present.
1643 Channel Event Logging
1644 ---------------------
1645 * A new interface, CEL, is introduced here. CEL logs single events, much like
1646 the AMI, but it differs from the AMI in that it logs to db backends much
1647 like CDR does; is based on the event subsystem introduced by Russell, and
1648 can share in all its benefits; allows multiple backends to operate like CDR;
1649 is specialized to event data that would be of concern to billing sytems,
1650 like CDR. Backends for logging and accounting calls have been produced,
1651 but a new CDR backend is still in development.
1655 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
1656 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
1657 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
1658 * Multiple files and formats can now be specified in cdr_custom.conf.
1659 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
1660 See configs/cdr_syslog.conf.sample for more information.
1661 * A 'sequence' field has been added to CDRs which can be combined with
1662 linkedid or uniqueid to uniquely identify a CDR.
1663 * Handling of billsec and duration field has changed. If your table definition
1664 specifies those fields as float,double or similar they will now be logged with
1665 microsecond accuracy instead of a whole integer.
1667 Calendaring for Asterisk
1668 ------------------------
1669 * A new set of modules were added supporing calendar integration with Asterisk.
1670 Dialplan functions for reading from and writing to calendars are included,
1671 as well as the ability to execute dialplan logic upon calendar event notifications.
1672 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
1673 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
1674 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
1675 2003 support does not support forms-based authentication).
1677 Call Completion Supplementary Services for Asterisk
1678 ---------------------------------------------------
1679 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
1680 DAHDI/ISDN supports call completion for the following switch types:
1681 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
1682 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
1684 Multicast RTP Support
1685 ---------------------
1686 * A new RTP engine and channel driver have been added which supports Multicast RTP.
1687 The channel driver can be used with the Page application to perform multicast RTP
1688 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
1689 Type can be either basic or linksys.
1690 Destination is the IP address and port for the RTP packets.
1691 Control address is specific to the linksys type and is used for sending the control
1692 packets unique to them.
1694 Security Events Framework
1695 -------------------------
1696 * Asterisk has a new C API for reporting security events. The module res_security_log
1697 sends these events to the "security" logger level. Currently, AMI is the only
1698 Asterisk component that reports security events. However, SIP support will be
1699 coming soon. For more information on the security events framework, see the
1700 "Asterisk Security Framework" section of the Asterisk wiki at
1701 https://wiki.asterisk.org/wiki/x/wgBQ
1702 * SIP support was added in Asterisk 10
1703 * This API now supports IPv6 addresses
1707 * A technology independent fax frontend (res_fax) has been added to Asterisk.
1708 * A spandsp based fax backend (res_fax_spandsp) has been added.
1709 * The app_fax module has been deprecated in favor of the res_fax module and
1710 the new res_fax_spandsp backend.
1711 * The SendFAX and ReceiveFAX applications now send their log messages to a
1712 'fax' logger level, instead of to the generic logger levels. To see these
1713 messages, the system's logger.conf file will need to direct the 'fax' logger
1714 level to one or more destinations; the logger.conf.sample file includes an
1715 example of how to do this. Note that if the 'fax' logger level is *not*
1716 directed to at least one destination, log messages generated by these
1717 applications will be lost, and that if the 'fax' logger level is directed to
1718 the console, the 'core set verbose' and 'core set debug' CLI commands will
1719 have no effect on whether the messages appear on the console or not.
1723 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
1724 Now, in order to enable transmitting silence during record the transmit_silence
1725 option should be used. transmit_silence_during_record remains a valid option, but
1726 defaults to the behavior of the transmit_silence option.
1727 * Addition of the Unit Test Framework API for managing registration and execution
1728 of unit tests with the purpose of verifying the operation of C functions.
1729 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
1730 XMPP text messages to the remote JID.
1731 * Modules.conf has a new option - "require" - that marks a module as critical for
1732 the execution of Asterisk.
1733 If one of the required modules fail to load, Asterisk will exit with a return
1735 * An 'X' option has been added to the asterisk application which enables #exec support.
1736 This allows #exec to be used in asterisk.conf.
1737 * jabber.conf supports a new option auth_policy that toggles auto user registration.
1738 * A new lockconfdir option has been added to asterisk.conf to protect the
1739 configuration directory (/etc/asterisk by default) during reloads.
1740 * The parkeddynamic option has been added to features.conf to enable the creation
1741 of dynamic parkinglots.
1742 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
1743 the reportalarms config option.
1744 * chan_dahdi supports dialing configuring and dialing by device file name.
1745 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
1746 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
1747 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
1748 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
1749 Handy for the above name-based syntax as it does not depend on
1750 initialization order.
1751 * The Realtime dialplan switch now caches entries for 1 second. This provides a
1752 significant increase in performance (about 3X) for installations using this switchtype.
1753 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
1754 AIS. For more information, please see the Distributed Device State section of the
1755 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1756 * The addition of G.719 pass-through support.
1757 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
1758 during device configuration.
1759 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
1760 have less than 3 lines on the LCD.
1761 * Realtime now supports database failover. See the sample extconfig.conf for details.
1762 * The addition of improved translation path building for wideband codecs. Sample
1763 rate changes during translation are now avoided unless absolutely necessary.
1764 * The addition of the res_stun_monitor module for monitoring and reacting to network
1765 changes while behind a NAT.
1766 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
1767 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
1768 These allow support for any Administration. Default is AT&T values.
1772 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
1773 optionally accept a filename, to apply the setting only to the code generated from
1774 that source file when Asterisk was built. However, there are some modules in Asterisk
1775 that are composed of multiple source files, so this did not result in the behavior
1776 that users expected. In this version, 'core set debug' and 'core set verbose'
1777 can optionally accept *module* names instead (with or without the .so extension),
1778 which applies the setting to the entire module specified, regardless of which source
1779 files it was built from.
1780 * New 'manager show settings' command showing the current settings loaded from
1782 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
1783 the channel hangup request to all channels.
1784 * Added a "core reload" CLI command that executes a global reload of Asterisk.
1786 ------------------------------------------------------------------------------
1787 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
1788 ------------------------------------------------------------------------------
1792 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
1793 Snom phones use this for call pickup of extensions that the phone is
1795 * Added support for setting the domain in the URI for caller of an
1796 outbound call by using the SIPFROMDOMAIN channel variable.
1797 * Added a new configuration option "remotesecret" for authentication to
1798 remote services. For backwards compatibility, "secret" still has the
1799 same function as before, but now you can configure both a remote secret and a
1800 local secret for mutual authentication.
1801 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
1802 the sound will be played to the target of an attended transfer
1803 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
1804 finer control over how many peers Asterisk will qualify and the gap between them
1805 when all peers need to be qualified at the same time.
1806 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
1807 (either globally or for a specific peer), chan_sip will treat any SDP data
1808 it receives as new data and update the media stream accordingly. By
1809 default, Asterisk will only modify the media stream if the SDP session
1810 version received is different from the current SDP session version. This
1811 option is required to interoperate with devices that have non-standard SDP
1812 session version implementations (observed with Microsoft OCS). This option
1813 is disabled by default.
1814 * The parsing of register => lines in sip.conf has been modified to allow a port
1815 to be present in the "user" portion. Please see the sip.conf.sample file for more
1817 * Added support for subscribing to MWI on a remote server and making the status available
1818 as a mailbox. Please see the sip.conf.sample file for more information.
1819 * Added a function to remove SIP headers added in the dialplan before the
1820 first INVITE is generated - SIPRemoveHeader()
1821 * Channel variables set with setvar= in a device configuration is now
1822 set both for inbound and outbound calls.
1823 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
1827 * Added immediate option to iax.conf
1828 * Added forceencryption option to iax.conf
1829 * Added Encryption and Trunk status to manager command "iaxpeers"
1833 * The configuration file now holds separate sections for devices and lines.
1834 Please have a look at configs/skinny.conf.sample and change your skinny.conf
1839 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
1840 support for LibOpenR2. http://www.libopenr2.org/
1841 * The UK option waitfordialtone has been added for use with BT analog
1843 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
1844 is used in conjunction with the 'faxdetect' configuration option. When
1845 'faxbuffers' is used and fax tones are detected, the channel will dynamically
1846 switch to the configured faxbuffers policy. For example, to use 6 buffers
1847 and a 'full' buffer policy for a fax transmission, add:
1849 The faxbuffers configuration will be in affect until the call is torn down.
1850 * Added service message support for 4ESS/5ESS switches.
1854 * For DAHDI channels, the CHANNEL() dialplan function now
1855 supports changing the channel's buffer policy (for the current
1856 call only), using this syntax:
1858 exten => s,n,Set(CHANNEL(buffers)=6,full)
1860 This would change the channel to the 'full' buffer policy and
1861 6 (six) buffers. Possible options for this setting are the same
1862 as those in chan_dahdi.conf.
1863 * Added a new dialplan function, CURLOPT, which permits setting various
1864 options that may be useful with the CURL dialplan function, such as
1865 cookies, proxies, connection timeouts, passwords, etc.
1866 * Permit the syntax and synopsis fields of the corresponding dialplan
1867 functions to be individually set from func_odbc.conf.
1868 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
1869 * func_odbc now may specify an insert query to execute, when the write query
1870 affects 0 rows (usually indicating that no such row exists).
1871 * Added a new dialplan function, LISTFILTER, which permits removing elements
1872 from a set list, by name. Uses the same general syntax as the existing CUT
1873 and FIELDQTY dialplan functions, which also manage lists.
1874 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
1875 obtaining realtime data from the dialplan.
1876 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
1877 a subroutine when using the GoSub() and Return() applications.
1878 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
1879 of "core show function AUDIOHOOK_INHERIT" from the CLI
1880 * Added AES_ENCRYPT. For information on its use, please see the output
1881 of "core show function AES_ENCRYPT" from the CLI
1882 * Added AES_DECRYPT. For information on its use, please see the output
1883 of "core show function AES_DECRYPT" from the CLI
1884 * func_odbc now supports database transactions across multiple queries.
1888 * Scheduled meetme conferences may now have their end times extended by
1890 * app_authenticate now gives the ability to select a prompt other than
1892 * app_directory now pays attention to the searchcontexts setting in
1893 voicemail.conf and will look through all contexts, if no context is
1894 specified in the initial argument.
1895 * A new application, Originate, has been introduced, that allows asynchronous
1896 call origination from the dialplan.
1897 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
1898 in addition to the setting in the "general" context.
1899 * Added ConfBridge dialplan application which does conference bridges without
1900 DAHDI. For information on its use, please see the output of
1901 "core show application ConfBridge" from the CLI.
1905 * The Asterisk CLI has a new command, "channel redirect", which is similar in
1906 operation to the AMI Redirect action.
1907 * extensions.conf now allows you to use keyword "same" to define an extension
1908 without actually specifying an extension. It uses exactly the same pattern
1909 as previously used on the last "exten" line. For example:
1910 exten => 123,1,NoOp(something)
1911 same => n,SomethingElse()
1912 * musiconhold.conf classes of type 'files' can now use relative directory paths,
1913 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
1914 * All deprecated CLI commands are removed from the sourcecode. They are now handled
1915 by the new clialiases module. See cli_aliases.conf.sample file.
1916 * Times within timespecs are now accurate down to the minute. This is a change
1917 from historical Asterisk, which only provided timespecs rounded to the nearest
1918 even (read: evenly divisible by 2) minute mark.
1919 * The realtime switch now supports an option flag, 'p', which disables searches for
1921 * In addition to a time range and date range, timespecs now accept a 5th optional
1922 argument, timezone. This allows you to perform time checks on alternate
1923 timezones, especially if those daylight savings time ranges vary from your
1924 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
1926 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
1927 give you the correct output for an asterisk box behind nat. It will give you the
1928 externhost and localnet settings.
1929 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
1930 can connect calls in passthrough mode, as well as record and play back files.
1931 * Successful and unsuccessful call pickup can now be alerted through sounds, by
1932 using pickupsound and pickupfailsound in features.conf.
1933 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
1934 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
1935 instead of the /var/run/asterisk.pid where it used to be. This will make
1936 installs as non-root easier to manage.
1941 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
1942 be written; they will no longer be explicitly written.
1944 Asterisk Manager Interface
1945 --------------------------
1946 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
1947 a non-empty value) in your request. If you do this, any pending AMI events will
1948 *not* be included in the response to your request as they would normally, but
1949 will be left in the event queue for the next request you make to retrieve. For
1950 some applications, this will allow you to guarantee that you will only see
1951 events in responses to 'WaitEvent' actions, and can better know when to expect them.
1952 To know whether the Asterisk server supports this header or not, your client can
1953 inspect the first response back from the server to see if it includes this header:
1955 Pragma: SuppressEvents
1957 If this is included, the server supports event suppression.
1959 * Added 4 new Actions to list skinny device(s) and line(s)
1965 LDAP Schema File Additions
1966 --------------------------
1967 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
1968 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
1970 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
1971 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
1972 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
1973 * Removed redundant IPaddr (there's already IPAddress)
1974 - Gives more configuration Flags for SIP-Users available (tested)
1975 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
1976 without extensibleObject (which really should be the last resort); gives
1977 also additional possibilities for LDAP-filter
1979 ------------------------------------------------------------------------------
1980 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
1981 ------------------------------------------------------------------------------
1983 Device State Handling
1984 ---------------------
1985 * The event infrastructure in Asterisk got another big update to help support
1986 distributed events. It currently supports distributed device state and
1987 distributed Voicemail MWI (Message Waiting Indication). A new module has
1988 been merged, res_ais, which facilitates communicating events between servers.
1989 It uses the SAForum AIS (Service Availability Forum Application Interface
1990 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
1991 a cluster of Asterisk servers, and to share events between them. For more
1992 information on setting this up, refer to the Distributed Device State section
1993 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1997 * Added a new dialplan function, AST_CONFIG(), which allows you to access
1998 variables from an Asterisk configuration file.
1999 * The JACK_HOOK function now has a c() option to supply a custom client name.
2000 * Added two new dialplan functions from libspeex for audio gain control and
2001 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
2002 rx directions of a channel from the dialplan.
2003 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
2004 based on other parameters. The default is still to search based on the
2005 forwarding station ID. However, there are new options that allow you to search
2006 based on the message desk terminal ID, or the message desk number.
2007 * TIMEOUT() has been modified to be accurate down to the millisecond.
2008 * ENUM*() functions now include the following new options:
2009 - 'u' returns the full URI and does not strip off the URI-scheme.
2010 - 's' triggers ISN specific rewriting
2011 - 'i' looks for branches into an Infrastructure ENUM tree
2012 - 'd' for a direct DNS lookup without any flipping of digits.
2013 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
2014 * CHANNEL() now has options for the maximum, minimum, and standard or normal
2015 deviation of jitter, rtt, and loss for a call using chan_sip.
2017 DAHDI channel driver (chan_dahdi) Changes
2018 ----------------------------------------
2019 * Channels can now be configured using named sections in chan_dahdi.conf, just
2020 like other channel drivers, including the use of templates.
2021 * The default for pridialplan has changed from 'national' to 'unknown'.
2025 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
2026 to something that matches the pattern a hint will be created using the contents
2027 and variables evaluated.
2028 * Dialplan matching has been extended to allow an extension to return to the
2029 PBX core to wait for more digits. This is done by using the new dialplan
2030 application called "Incomplete". This will permit a whole new level of
2031 extension control, by giving the administrator more control over early
2032 matches employing one of the short-circuit pattern match operators. Note
2033 that custom applications can trigger this same behavior by returning the
2034 special value AST_PBX_INCOMPLETE.
2038 * Directory now permits both first and last names to be matched at the same
2039 time. In addition, the number of digits to enter of the name can be set in
2040 the arguments to Directory; previously, you could enter only 3, regardless
2041 of how many names are in your company. For large companies, this should be
2043 * Voicemail now permits a mailbox setting to wrap around from first to last
2044 messages, if the "messagewrap" option is set to a true value.
2045 * Voicemail now permits an external script to be run, for password validation.
2046 The script should output "VALID" or "INVALID" on stdout, depending upon the
2047 wish to validate or invalidate the password given. Arguments are:
2048 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
2050 * Dial has a new option: F(context^extension^pri), which permits a callee to
2051 continue in the dialplan, at the specified label, if the caller hangs up.
2052 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
2053 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
2054 * The Jack application now has a c() option to supply a custom client name.
2055 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
2056 like the pre-existing whisper mode, except that the spy can also talk to the
2057 participant on the bridged channel as well.
2058 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
2059 to be spoken instead of the channel name or number. For more information on the
2060 use of this option, issue the command "core show application ChanSpy" from the
2062 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
2063 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
2064 words, if using the 'd' option, it is not possible to enter a number to append to
2065 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
2066 change to whisper mode, and pressing 6 will change to barge mode.
2067 * ExternalIVR now takes several options that affect the way it performs, as
2068 well as having several new commands. Please see the External IVR page on the Asterisk
2069 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
2070 * Added ability to communicate over a TCP socket instead of forking a child process for the
2071 ExternalIVR application.
2072 * ChanIsAvail has a new option, 'a', which will return all available channels instead
2073 of just the first one if you give the function more then one channel to check.
2074 * PrivacyManager now takes an option where you can specify a context where the
2075 given number will be matched. This way you have more control over who is allowed
2076 and it stops the people who blindly enter 10 digits.
2077 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
2078 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
2079 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
2080 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
2081 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
2082 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
2083 * The Dial() application no longer copies the language used by the caller to the callee's
2084 channel. If you desire for the caller's channel's language to be used for file playback
2085 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
2086 * SendImage() no longer hangs up the channel on error; instead, it sets the
2087 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
2088 'UNSUPPORTED'. This change makes SendImage() more consistent with other
2090 * Park has a new option, 's', which silences the announcement of the parking space number.
2091 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
2092 invalid input and will be assumed to mean that no timeout is desired.
2096 * Added DNS manager support to registrations for peers referencing peer entries.
2097 DNS manager runs in the background which allows DNS lookups to be run asynchronously
2098 as well as periodically updating the IP address. These properties allow for
2099 better performance as well as recovery in the event of an IP change.
2100 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
2101 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
2102 These changes also provide performance improvements for call setup and tear down.
2103 * Added ability to specify registration expiry time on a per registration basis in
2105 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
2107 * Added t38pt_usertpsource option. See sip.conf.sample for details.
2108 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
2109 * 'sip show peers' and 'sip show users' display their entries sorted in
2110 alphabetical order, as opposed to the order they were in, in the config
2112 * Videosupport now supports an additional option, "always", which always sets
2113 up video RTP ports, even on clients that don't support it. This helps with
2114 callfiles and certain transfers to ensure that if two video phones are
2115 connected, they will always share video feeds.
2119 * Existing DNS manager lookups extended to check for SRV records.
2120 * IAX2 encryption support has been improved to support periodic key rotation
2121 within a call for enhanced security. The option "keyrotate" has been
2122 provided to disable this functionality to preserve backwards compatibility
2123 with older versions of IAX2 that do not support key rotation.
2127 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
2128 data tree based on the given <path>.
2129 * New CLI command "data show providers" that will display all the registered
2131 * New CLI command, "config reload <file.conf>" which reloads any module that
2132 references that particular configuration file. Also added "config list"
2133 which shows which configuration files are in use.
2134 * New CLI commands, "pri show version" and "ss7 show version" that will
2135 display which version of libpri and libss7 are being used, respectively.
2136 A new API call was added so trunk will now have to be compiled against
2137 a versions of libpri and libss7 that have them or it will not know that
2138 these libraries exist.
2139 * The commands "core show globals", "core set global" and "core set chanvar" has
2140 been deprecated in favor of the more semanticly correct "dialplan show globals",
2141 "dialplan set chanvar" and "dialplan set global".
2142 * New CLI command "dialplan show chanvar" to list all variables associated
2143 with a given channel.
2147 * Addresses managed by DNS manager now can check to see if there is a DNS
2148 SRV record for a given domain and will use that hostname/port if present.
2150 AMI - The manager (TCP/TLS/HTTP)
2151 --------------------------------
2152 * The Status command now takes an optional list of variables to display
2153 along with channel status.
2154 * The QueueEntry event now also includes the channel's uniqueid
2158 * res_odbc no longer has a limit of 1023 total possible unshared connections,
2159 as some people were running into this limit. This limit has been increased
2164 * The TRANSFER queue log entry now includes the the caller's original
2165 position in the transferred-from queue.
2166 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
2167 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
2168 as well as an explanation about timeout options in general
2169 * Added a new option - C - for forcing the "answered elsewhere" flag on
2170 cancellation of calls in to members of the queue. This is to avoid the
2171 call to a member of a queue having the call listed as a "missed call".
2175 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
2176 adaptive capabilities. What this means in practical terms is that if your
2177 realtime table lacks critical fields, Asterisk will now emit warnings to
2178 that effect. Also, some of the realtime drivers have the ability (if
2179 configured) to automatically add those columns to the table with the
2180 correct type and length.
2184 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
2185 the 'setvar' option to cause a given audio file to be played upon completion
2186 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
2187 Skinny channels only.
2188 * You can now compile Asterisk against the Hoard Memory Allocator, see the
2189 Hoard page on the Asterisk wiki for more information:
2190 https://wiki.asterisk.org/wiki/x/pQBB
2191 * Config file variables may now be appended to, by using the '+=' append
2192 operator. This is most helpful when working with long SQL queries in
2193 func_odbc.conf, as the queries no longer need to be specified on a single
2195 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
2196 which will add a second to the billsec when the ending
2197 time is set, if the number in the microseconds field of the end time is
2198 greater than the number of microseconds in the answer time. This allows
2199 users to count the 'initiated' seconds in their billing records.
2201 ------------------------------------------------------------------------------
2202 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
2203 ------------------------------------------------------------------------------
2205 AMI - The manager (TCP/TLS/HTTP)
2206 --------------------------------
2207 * Manager has undergone a lot of changes, all of them documented
2208 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
2209 * Manager version has changed to 1.1
2210 * Added a new action 'CoreShowChannels' to list currently defined channels
2211 and some information about them.
2212 * Added a new action 'SIPshowregistry' to list SIP registrations.
2213 * Added TLS support for the manager interface and HTTP server
2214 * Added the URI redirect option for the built-in HTTP server
2215 * The output of CallerID in Manager events is now more consistent.
2216 CallerIDNum is used for number and CallerIDName for name.
2217 * Enable https support for builtin web server.
2218 See configs/http.conf.sample for details.
2219 * Added a new action, GetConfigJSON, which can return the contents of an
2220 Asterisk configuration file in JSON format. This is intended to help
2221 improve the performance of AJAX applications using the manager interface
2223 * SIP and IAX manager events now use "ChannelType" in all cases where we
2224 indicate channel driver. Previously, we used a mixture of "Channel"
2225 and "ChannelDriver" headers.
2226 * Added a "Bridge" action which allows you to bridge any two channels that
2227 are currently active on the system.
2228 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
2229 the voicemail users setup.
2230 * Added 'DBDel' and 'DBDelTree' manager commands.
2231 * cdr_manager now reports events via the "cdr" level, separating it from
2232 the very verbose "call" level.
2233 * Manager users are now stored in memory. If you change the manager account
2234 list (delete or add accounts) you need to reload manager.
2235 * Added Masquerade manager event for when a masquerade happens between
2237 * Added "manager reload" command for the CLI
2238 * Lots of commands that only provided information are now allowed under the
2239 Reporting privilege, instead of only under Call or System.
2240 * The IAX* commands now require either System or Reporting privilege, to
2241 mirror the privileges of the SIP* commands.
2242 * Added ability to retrieve list of categories in a config file.
2243 * Added ability to retrieve the content of a particular category.
2244 * Added ability to empty a context.
2245 * Created new action to create a new file.
2246 * Updated delete action to allow deletion by line number with respect to category.
2247 * Added new action insert to add new variable to category at specified line.
2248 * Updated action newcat to allow new category to be inserted in file above another
2250 * Added new event "JitterBufStats" in the IAX2 channel
2251 * Originate now requires the Originate privilege and, if you want to call out
2252 to a subshell, it requires the System privilege, as well. This was done to
2253 enhance manager security.
2254 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
2255 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
2256 or manager show command Atxfer from the CLI
2257 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
2258 details or manager show command IAXregistry from the CLI
2262 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
2263 state in the dialplan, as well as creating custom device states that are
2264 controllable from the dialplan.
2265 * Extend CALLERID() function with "pres" and "ton" parameters to
2266 fetch string representation of calling number presentation indicator
2267 and numeric representation of type of calling number value.
2268 * MailboxExists converted to dialplan function
2269 * A new option to Dial() for telling IP phones not to count the call
2270 as "missed" when dial times out and cancels.
2271 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
2272 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
2273 held for any given channel. Also, locks are automatically freed when a
2275 * Added HINT() dialplan function that allows retrieving hint information.
2276 Hints are mappings between extensions and devices for the sake of
2277 determining the state of an extension. This function can retrieve the list
2278 of devices or the name associated with a hint.
2279 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
2281 * Added SYSINFO() dialplan function which allows retrieval of system information
2282 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
2283 the existence of a dialplan target.
2284 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
2285 upper and lower case, respectively.
2286 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
2287 ID for the call (not the Asterisk call ID or unique ID), provided that the
2288 channel driver supports this. For SIP, you get the SIP call-ID for the
2289 bridged channel which you can store in the CDR with a custom field.
2293 * Added CLI permissions, config file: cli_permissions.conf
2294 default is to allow all commands for every local user/group.
2295 Also this new feature added three new CLI commands:
2296 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
2297 - cli reload permissions
2298 - cli show permissions
2299 * New CLI command "core show hint" (usage: core show hint <exten>)
2300 * New CLI command "core show settings"
2301 * Added 'core show channels count' CLI command.
2302 * Added the ability to set the core debug and verbose values on a per-file basis.
2303 * Added 'queue pause member' and 'queue unpause member' CLI commands
2304 * Ability to set process limits ("ulimit") without restarting Asterisk
2305 * Enhanced "agi debug" to print the channel name as a prefix to the debug
2306 output to make debugging on busy systems much easier.
2307 * New CLI commands "dialplan set extenpatternmatching true/false"
2308 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
2309 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
2310 listed in the startup_commands section of cli.conf will get executed.
2311 * Added a CLI command, "devstate change", which allows you to set custom device
2312 states from the func_devstate module that provides the DEVICE_STATE() function
2313 and handling of the "Custom:" devices.
2314 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
2315 sorted into the different possible callbacks, with the number of entries
2316 currently scheduled for each. Gives you a feel for how busy the sip channel
2318 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
2319 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
2320 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
2324 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
2325 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
2326 for a received call. If it is detected, the channel will jump to the
2327 'fax' extension in the dialplan.
2328 * The default SIP useragent= identifier now includes the Asterisk version
2329 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
2330 If set, and the incoming request carries authentication info,
2331 the username to match in the users list is taken from the Digest header
2332 rather than from the From: field. This feature is considered experimental.
2333 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
2334 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
2335 * The "localmask" setting was removed in version 1.2 and the reminder about it
2336 being removed is now also removed.
2337 * A new option "busylevel" for setting a level of calls where asterisk reports
2338 a device as busy, to separate it from call-limit. This value is also added
2339 to the SIP_PEER dialplan function.
2340 * A new realtime family called "sipregs" is now supported to store SIP registration
2341 data. If this family is defined, "sippeers" will be used for configuration and
2342 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
2343 registration data, as before.
2344 * The SIPPEER function have new options for port address, call and pickup groups
2345 * Added support for T.140 realtime text in SIP/RTP
2346 * The "checkmwi" option has been removed from sip.conf, as it is no longer
2347 required due to the restructuring of how MWI is handled. See the descriptions
2348 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
2349 for more information.
2350 * Added rtpdest option to CHANNEL() dialplan function.
2351 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
2352 * SIP now adds a header to the CANCEL if the call was answered by another phone
2353 in the same dial command, or if the new c option in dial() is used.
2354 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
2355 states it is not needed. For phones, however, that do require it the "registertrying" option
2356 has been added so it can be enabled.
2357 * A new option called "callcounter" (global/peer/user level) enables call counters needed
2358 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
2359 used to enable this functionality).
2360 * New settings for timer T1 and timer B on a global level or per device. This makes it
2361 possible to force timeout faster on non-responsive SIP servers. These settings are
2362 considered advanced, so don't use them unless you have a problem.
2363 * Added a dial string option to be able to set the To: header in an INVITE to any
2365 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
2366 the qualify frequency.
2367 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
2368 were not properly torn down due to network or endpoint failures during an established
2370 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
2371 and configs/sip.conf.sample for more information on how it is used.
2372 * Added a new configuration option "authfailureevents" that enables manager events when
2373 a peer can't authenticate properly.
2374 * Added DNS manager support to registrations for peers not referencing a peer entry.
2378 * Added the trunkmaxsize configuration option to chan_iax2.
2379 * Added the srvlookup option to iax.conf
2380 * Added support for OSP. The token is set and retrieved through the CHANNEL()
2383 XMPP Google Talk/Jingle changes
2384 -------------------------------
2385 * Added the bindaddr option to gtalk.conf.
2389 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
2390 * Proper codec support in chan_skinny.
2391 * Added settings for IP and Ethernet QoS requests
2395 * Added separate settings for media QoS in mgcp.conf
2397 Console Channel Driver changes
2398 ------------------------------
2399 * Added experimental support for video send & receive to chan_oss.
2400 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
2403 Phone channel changes (chan_phone)
2404 ----------------------------------
2405 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
2407 H.323 channel Changes
2408 ---------------------
2409 * H323 remote hold notification support added (by NOTIFY message
2410 and/or H.450 supplementary service)
2412 Local channel changes
2413 ---------------------
2414 * The device state functionality in the Local channel driver has been updated
2415 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
2416 to just UNKNOWN if the extension exists.
2417 * Added jitterbuffer support for chan_local. This allows you to use the
2418 generic jitterbuffer on incoming calls going to Asterisk applications.
2419 For example, this would allow you to use a jitterbuffer for an incoming
2420 SIP call to Voicemail by putting a Local channel in the middle. This
2421 feature is enabled by using the 'j' option in the Dial string to the Local
2422 channel in conjunction with the existing 'n' option for local channels.
2423 * A 'b' option has been added which causes chan_local to return the actual channel
2424 that is behind it when queried. This is useful for transfer scenarios as the
2425 actual channel will be transferred, not the Local channel.
2427 Agent channel changes
2428 ----------------------
2429 * The ackcall and endcall options are now supplemented with options acceptdtmf
2430 and enddtmf. These allow for the DTMF keypress to be configurable. The options
2431 default to their old hard-coded values ('#' and '*' respectively) so this should
2432 not break any existing agent installations.
2434 DAHDI channel driver (chan_dahdi) Changes
2435 ----------------------------------------
2436 * SS7 support (via libss7 library)
2437 * In India, some carriers transmit CID via dtmf. Some code has been added
2438 that will handle some situations. The cidstart=polarity_IN choice has been added for
2439 those carriers that transmit CID via dtmf after a polarity change.
2440 * CID matching information is now shown when doing 'dialplan show'.
2441 * Added dahdi show version CLI command.
2442 * Added setvar support to chan_dahdi.conf channel entries.
2443 * Added two new options: mwimonitor and mwimonitornotify. These options allow
2444 you to enable MWI monitoring on FXO lines. When the MWI state changes,
2445 the script specified in the mwimonitornotify option is executed. An internal
2446 event indicating the new state of the mailbox is also generated, so that
2447 the normal MWI facilities in Asterisk work as usual.
2448 * Added signalling type 'auto', which attempts to use the same signalling type
2449 for a channel as configured in DAHDI. This is primarily designed for analog
2450 ports, but will also work for digital ports that are configured for FXS or FXO
2451 signalling types. This mode is also the default now, so if your chan_dahdi.conf
2452 does not specify signalling for a channel (which is unlikely as the sample
2453 configuration file has always recommended specifying it for every channel) then
2454 the 'auto' mode will be used for that channel if possible.
2455 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
2456 state for a channel; also ensured that the DNDState Manager event is
2457 emitted no matter how the DND state is set or cleared.
2461 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
2462 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
2463 for details. This new channel driver allows you to use Nortel i2002,
2464 i2004, and i2050 phones with Asterisk.
2465 * Added a new channel driver, chan_console, which uses portaudio as a cross
2466 platform audio interface. It was written as a channel driver that would
2467 work with Mac CoreAudio, but portaudio supports a number of other audio
2468 interfaces, as well. Note that this channel driver requires v19 or higher
2469 of portaudio; older versions have a different API.
2473 * Added the ability to specify arguments to the Dial application when using
2474 the DUNDi switch in the dialplan.
2475 * Added the ability to set weights for responses dynamically. This can be
2476 done using a global variable or a dialplan function. Using the SHELL()
2477 function would allow you to have an external script set the weight for
2479 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
2480 functions will allow you to initiate a DUNDi query from the dialplan,
2481 find out how many results there are, and access each one.
2482 * Added the ability to specifiy a port for a dundi peer.
2486 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
2487 functions will allow you to initiate an ENUM lookup from the dialplan,
2488 and Asterisk will cache the results. ENUMRESULT can be used to access
2489 the results without doing multiple DNS queries.
2493 * Added the ability to customize which sound files are used for some of the
2494 prompts within the Voicemail application by changing them in voicemail.conf
2495 * Added the ability for the "voicemail show users" CLI command to show users
2496 configured by the dynamic realtime configuration method.
2497 * MWI (Message Waiting Indication) handling has been significantly
2498 restructured internally to Asterisk. It is now totally event based
2499 instead of polling based. The voicemail application will notify other
2500 modules that have subscribed to MWI events when something in the mailbox
2502 This also means that if any other entity outside of Asterisk is changing
2503 the contents of mailboxes, then the voicemail application still needs to
2504 poll for changes. Examples of situations that would require this option
2505 are web interfaces to voicemail or an email client in the case of using
2506 IMAP storage. So, two new options have been added to voicemail.conf
2507 to account for this: "pollmailboxes" and "pollfreq". See the sample
2508 configuration file for details.
2509 * Added "tw" language support
2510 * Added support for storage of greetings using an IMAP server
2511 * Added ability to customize forward, reverse, stop, and pause keys for message playback
2512 * SMDI is now enabled in voicemail using the smdienable option.
2513 * A "lockmode" option has been added to asterisk.conf to configure the file
2514 locking method used for voicemail, and potentially other things in the
2515 future. The default is the old behavior, lockfile. However, there is a
2516 new method, "flock", that uses a different method for situations where the
2517 lockfile will not work, such as on SMB/CIFS mounts.
2518 * Added the ability to backup deleted messages, to ease recovery in the case
2519 that a user accidentally deletes a message, and discovers that they need it.
2520 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
2521 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
2522 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
2523 voicemail boxes. The SMDI interface can also poll for MWI changes when some
2524 outside entity is modifying the state of the mailbox (such as IMAP storage or
2525 a web interface of some kind).
2526 * Added the support for marking messages as "urgent." There are two methods to accomplish
2527 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
2528 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
2529 the message as urgent after he has recorded a voicemail by following the voice instructions.
2530 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
2535 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
2536 used across multiple queues.
2537 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
2538 setqueueentryvar options for each queue, see queues.conf.sample for details.
2539 * Added keepstats option to queues.conf which will keep queue
2540 statistics during a reload.
2541 * setinterfacevar option in queues.conf also now sets a variable
2542 called MEMBERNAME which contains the member's name.
2543 * Added 'Strategy' field to manager event QueueParams which represents
2544 the queue strategy in use.
2545 * Added option to run macro when a queue member is connected to a caller,
2546 see queues.conf.sample for details.
2547 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
2548 does not count paused queue members as unavailable.
2549 * Added min-announce-frequency option to queues.conf which allows you to control the
2550 minimum amount of time between queue announcements for use when the caller's queue
2551 position changes frequently.
2552 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
2554 * Added ability for non-realtime queues to have realtime members
2555 * Added the "linear" strategy to queues.
2556 * Added the "wrandom" strategy to queues.
2557 * Added new channel variable QUEUE_MIN_PENALTY
2558 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
2559 rules in queuerules.conf. See configs/queuerules.conf.sample for details
2560 * Added a new parameter for member definition, called state_interface. This may be
2561 used so that a member may be called via one interface but have a different interface's
2562 device state reported.
2563 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
2564 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
2565 "manager show command QueueReset."
2566 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
2567 specified by the periodic-announce option, then one will be chosen randomly when it is time
2568 to play a periodic announcment
2569 * New configuration options: announce-position now takes two more values in addition to "yes" and
2570 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
2571 announce-position-limit. By setting announce-position to "limit" callers will only have their
2572 position announced if their position is less than what is specified by announce-position-limit.
2573 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
2574 will be told that their are more than announce-position-limit callers waiting.
2575 * Two new queue log events have been added. An ADDMEMBER event will be logged
2576 when a realtime queue member is added and a REMOVEMEMBER event will be logged
2577 when a realtime queue member is removed. Since there is no calling channel associated
2578 with these events, the string "REALTIME" is placed where the channel's unique id
2579 is typically placed.
2580 * The configuration method for the "joinempty" and "leavewhenempty" options has
2581 changed to a comma-separated list of methods of determining member availability
2582 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
2583 values are still accepted for backwards-compatibility, though.
2584 * The average talktime is now calculated on queues. This information is reported via the
2585 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
2586 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
2591 * The 'o' option to provide an optimization has been removed and its functionality
2592 has been enabled by default.
2593 * When a conference is created, the UNIQUEID of the channel that caused it to be
2594 created is stored. Then, every channel that joins the conference will have the
2595 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
2596 callers that come and go from long standing conferences.
2597 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
2598 except it does operations on a channel by name, instead of number in a conference.
2599 This is a very useful feature in combination with the 'X' option to ChanSpy.
2600 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
2602 * Added new RealTime functionality to provide support for scheduled conferencing.
2603 This includes optional messages to the caller if they attempt to join before
2604 the schedule start time, or to allow the caller to join the conference early.
2605 Also included is optional support for limiting the number of callers per
2606 RealTime conference.
2607 * Added the S() and L() options to the MeetMe application. These are pretty
2608 much identical to the S() and L() options to Dial(). They let you set
2609 timeouts for the conference, as well as have warning sounds played to
2610 let the caller know how much time is left, and when it is running out.
2611 * Added the ability to do "meetme concise" with the "meetme" CLI command.
2612 This extends the concise capabilities of this CLI command to include
2613 listing all conferences, instead of an addition to the other sub commands
2614 for the "meetme" command.
2615 * Added the ability to specify the music on hold class used to play into the
2616 conference when there is only one member and the M option is used.
2617 * Added MEETME_INFO dialplan function which provides a way to query
2618 various properties of a Meetme conference.
2619 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
2620 and *84: record in-conf
2622 Other Dialplan Application Changes
2623 ----------------------------------
2624 * Argument support for Gosub application
2625 * From the to-do lists: straighten out the app timeout args:
2626 Wait() app now really does 0.3 seconds- was truncating arg to an int.
2627 WaitExten() same as Wait().
2628 Congestion() - Now takes floating pt. argument.
2629 Busy() - now takes floating pt. argument.
2630 Read() - timeout now can be floating pt.
2631 WaitForRing() now takes floating pt timeout arg.
2632 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
2633 * Added 's' option to Page application.
2634 * Added an optional timeout argument to the Page application.
2635 * Added 'E', 'V', and 'P' commands to ExternalIVR.
2636 * Added 'o' and 'X' options to Chanspy.
2637 * Added a new dialplan application, Bridge, which allows you to bridge the
2638 calling channel to any other active channel on the system.
2639 * Added the ability to specify a music on hold class to play instead of ringing
2640 for the SLATrunk application.
2641 * The Read application no longer exits the dialplan on error. Instead, it sets
2642 READSTATUS to ERROR, which you can catch and handle separately.
2643 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
2644 of asking for verification of each name, one at a time.
2645 * Privacy() no longer uses privacy.conf, as all options are specifyable as
2646 direct options to the app.
2647 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
2649 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
2650 * The ChannelRedirect application no longer exits the dialplan if the given channel
2651 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
2652 or NOCHANNEL if the given channel was not found.
2653 * The silencethreshold setting that was previously configurable in multiple
2654 applications is now settable globally via dsp.conf.
2656 Music On Hold Changes
2657 ---------------------
2658 * A new option, "digit", has been added for music on hold classes in
2659 musiconhold.conf. If this is set for a music on hold class, a caller
2660 listening to music on hold can press this digit to switch to listening
2661 to this music on hold class.
2662 * Support for realtime music on hold has been added.
2663 * In conjunction with the realtime music on hold, a general section has
2664 been added to musiconhold.conf, its sole variable is cachertclasses. If this
2665 is set, then music on hold classes found in realtime will be cached in memory.
2669 * AEL upgraded to use the Gosub with Arguments instead
2670 of Macro application, to hopefully reduce the problems
2671 seen with the artificially low stack ceiling that
2672 Macro bumps into. Macros can only call other Macros
2673 to a depth of 7. Tests run using gosub, show depths
2674 limited only by virtual memory. A small test demonstrated
2675 recursive call depths of 100,000 without problems.
2676 -- in addition to this, all apps that allowed a macro
2677 to be called, as in Dial, queues, etc, are now allowing
2678 a gosub call in similar fashion.
2679 * AEL now generates LOCAL(argname) declarations when it
2680 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
2681 etc. That makes the arguments local in scope. The user
2682 can define their own local variables in macros, now,
2683 by saying "local myvar=someval;" or using Set() in this
2684 fashion: Set(LOCAL(myvar)=someval); ("local" is now
2686 * utils/conf2ael introduced. Will convert an extensions.conf
2687 file into extensions.ael. Very crude and unfinished, but
2688 will be improved as time goes by. Should be useful for a
2689 first pass at conversion.
2690 * aelparse will now read extensions.conf to see if a referenced
2691 macro or context is there before issueing a warning.
2692 * AEL parser sets a local channel variable ~~EXTEN~~, to
2693 preserve the value of ${EXTEN} thru switch statements.
2694 * New operator in $[...] expressions: the ~~ operator serves
2695 as a concatenation operator. AT THE MOMENT, it is really only
2696 necessary and useful in AEL, especially in if() expressions.
2697 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
2698 any enclosing double-quotes, and evaluate to the value of a
2699 concatenated with the value of b. For example if a is set to
2700 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
2701 evaluate to xyzabc .
2704 Call Features (res_features) Changes
2705 ------------------------------------
2706 * Added the parkedcalltransfers option to features.conf
2707 * Added parkedcallparking option to control one touch parking w/ parking
2709 * Added parkedcallhangup option to control disconnect feature w/ parking
2711 * Added parkedcallrecording option to control one-touch record w/ parking
2713 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
2714 parkedcalltransfers option support for multiple parking lots.
2715 * Added BRIDGE_FEATURES variable to set available features for a channel
2716 * The built-in method for doing attended transfers has been updated to
2717 include some new options that allow you to have the transferee sent
2718 back to the person that did the transfer if the transfer is not successful.
2719 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
2720 in features.conf.sample.
2721 * Added support for configuring named groups of custom call features in
2722 features.conf. This means that features can be written a single time, and
2723 then mapped into groups of features for different key mappings or easier
2725 * Updated the ParkedCall application to allow you to not specify a parking
2726 extension. If you don't specify a parking space to pick up, it will grab
2727 the first one available.
2728 * Added cli command 'features reload' to reload call features from features.conf
2729 * Moved into core asterisk binary.
2730 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
2731 * Added the ability for custom parking lots to be configured with their own
2732 parking extension with the parkext option.
2734 Language Support Changes
2735 ------------------------
2736 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
2737 * Added support for the Hungarian language for saying numbers, dates, and times.
2741 * Added SPEECH commands for speech recognition. A complete listing can be found
2743 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
2744 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
2745 does not behave as expected; the native command needs to be used, instead.
2746 * Added the ability to perform SRV lookups on fast AGI calls. To use this
2747 feature, simply use hagi: instead of agi: as the protocol portion
2748 of the URI parameter to the AGI function call in your dial plan. Also note
2749 that specifying a port number in the AGI URI will disable SRV lookups,
2750 even if you use the hagi: protocol.
2751 * No longer support MSG_OOB flag on HANGUP.
2755 * Added rotatestrategy option to logger.conf, along with two new options:
2756 "timestamp" which will use the time to name the logger files instead of
2757 sequence number; and "rotate", which rotates the names of the log files,
2758 similar to the way syslog rotates files.
2759 * Added exec_after_rotate option to logger.conf, which allows a system
2760 command to be run after rotation. This is primarily useful with
2761 rotatestrategy=rotate, to allow a limit on the number of log files kept
2762 and to ensure that the oldest log file gets deleted.
2763 * Added realtime support for the queue log
2767 * The cdr_manager module has a [mappings] feature, like cdr_custom,
2768 to add fields to the manager event from the CDR variables.
2769 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
2770 backend database CDR table. Specifically, additional, non-standard
2771 columns are supported, merely by setting the corresponding CDR variable in
2772 your dialplan. In addition, you may alias any column to another name (for
2773 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
2774 simply "alias src => ANI" in the configuration file). Records may be
2775 posted to more than one backend, simply by specifying multiple categories
2776 in the configuration file. And finally, you may filter which CDRs get
2777 posted to each backend, by specifying a filter (which the record must
2778 match) for the particular category. Filters are additive (meaning all
2779 rules must match to post that CDR).
2780 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
2781 module. Specifically, you may add additional columns into the table and
2782 they will be set, if you set the corresponding CDR variable name. Also,
2783 if you omit columns in your database table, they will be silently skipped
2784 (but a record will still be inserted, based on what columns remain). Note
2785 that the other two features from cdr_adaptive_odbc (alias and filter) are
2786 not currently supported.
2787 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
2788 has been disabled using the NoCDR application.
2790 Miscellaneous New Modules
2791 -------------------------
2792 * Added a new CDR module, cdr_sqlite3_custom.
2793 * Added a new realtime configuration module, res_config_sqlite
2794 * Added a new codec translation module, codec_resample, which re-samples
2795 signed linear audio between 8 kHz and 16 kHz to help support wideband
2797 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
2798 based on configuration templates that use Asterisk dialplan function and
2799 variable substitution. It should be possible to create phone profiles and
2800 templates that work for the majority of phones provisioned over http. It
2801 is currently only intended to provision a single user account per phone.
2802 An example profile and set of templates for Polycom phones is provided.
2803 NOTE: Polycom firmware is not included, but should be placed in
2804 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
2805 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
2806 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
2807 provided; there is a JACK() application, and a JACK_HOOK() function. Both
2808 interfaces create an input and output JACK port. The application makes
2809 these ports the endpoint of the call. The audio coming from the channel
2810 goes out the output port and whatever comes back in on the input port is
2811 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
2812 audiohook on the channel. This lets you run the audio coming from a
2813 channel through JACK, and whatever comes back in is what gets forwarded
2814 on as the channel's audio. This is very useful for building custom
2815 vocoders or doing recording or analysis of the channel's audio in another
2817 * Added a new module, res_config_curl, which permits using a HTTP POST url
2818 to retrieve, create, update, and delete realtime information from a remote
2819 web server. Note that this module requires func_curl.so to be loaded for
2820 backend functionality.
2821 * Added a new module, res_config_ldap, which permits the use of an LDAP
2822 server for realtime data access.
2823 * Added support for writing and running your dialplan in lua using the pbx_lua
2824 module. See configs/extensions.lua.sample for examples of how to do this.
2828 * Ability to use libcap to set high ToS bits when non-root
2829 on Linux. If configure is unable to find libcap then you
2830 can use --with-cap to specify the path.
2831 * Added maxfiles option to options section of asterisk.conf which allows you to specify
2832 what Asterisk should set as the maximum number of open files when it loads.
2833 * Added the jittertargetextra configuration option.
2834 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
2835 configuration files for the IP channel drivers. The new option is "cos".
2836 This information is also documented on the Asterisk wiki at
2837 https://wiki.asterisk.org/wiki/x/EYBG
2838 * When originating a call using AMI or pbx_spool that fails the reason for failure
2839 will now be available in the failed extension using the REASON dialplan variable.
2840 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
2841 It allows you to configure a prefix for auto-monitor recordings.
2842 * A new extension pattern matching algorithm, based on a trie, is introduced
2843 here, that could noticeably speed up mid-sized to large dialplans.
2844 It is NOT used by default, as duplicating the behaviour of the old pattern
2845 matcher is still under development. A config file option, in extensions.conf,
2846 in the [general] section, called "extenpatternmatchingnew", is by default
2847 set to false; setting that to true will force the use of the new algorithm.
2848 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
2849 be used to switch the algorithms at run time.
2850 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
2851 specifying which socket to use to connect to the running Asterisk daemon
2853 * Performance enhancements to the sched facility, which is used in
2854 the channel drivers, etc. Added hashtabs and doubly-linked lists
2855 to speed up deletion; start at the beginning or end of list to
2857 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
2858 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
2859 Added regression tests to the tests/ dir, also.
2860 * Added a refcount trace feature to astobj2 for those trying to balance
2861 object creation, deletion; work, play; space and time. See the
2862 notes in astobj2.h. Also, see utils/refcounter as well, as a
2863 quick way to find unbalanced refcounts in what could be a sea
2864 of objects that were balanced.
2865 * Added logging to 'make update' command. See update.log
2866 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
2867 do not come from the remote party.
2868 * Added the 'n' option to the SpeechBackground application to tell it to not
2869 answer the channel if it has not already been answered.
2870 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
2871 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
2873 * iLBC source code no longer included (see UPGRADE.txt for details)
2874 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
2875 deadlock is detected, a backtrace of the stack which led to the lock calls
2876 will be output to the CLI.
2877 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
2878 the "core show locks" CLI command will give lock information output as well
2879 as a backtrace of the stack which led to the lock calls.
2880 * users.conf now sports an optional alternateexts property, which permits
2881 allocation of additional extensions which will reach the specified user.
2882 * A new option for the configure script, --enable-internal-poll, has been added
2883 for use with systems which may have a buggy implementation of the poll system
2884 call. If you notice odd behavior such as the CLI being unresponsive on remote
2885 consoles, you may want to try using this option. This option is enabled by default
2886 on Darwin systems since it is known that the Darwin poll() implementation has
2890 --------------------
2891 * In addition to timing from DAHDI, there is a new timing module called
2892 res_timing_timerfd. In order to use this, you must be running Linux with
2893 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
2894 script will be able to tell if you have the requirements. From menuselect, select
2895 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob