1 ==============================================================================
3 === This file documents the new and/or enhanced functionality added in
4 === the Asterisk versions listed below. This file does NOT include
5 === changes in behavior that would not be backwards compatible with
6 === previous versions; for that information see the UPGRADE.txt file
7 === and the other UPGRADE files for older releases.
9 ==============================================================================
11 ------------------------------------------------------------------------------
12 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
13 ------------------------------------------------------------------------------
17 * The Asterisk build system will now build and install a shared library
18 (libasteriskssl.so) used to wrap various initialization and shutdown functions
19 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
20 that Asterisk can ensure that these functions do *not* get called by any
21 modules that are loaded into Asterisk, since they should only be called once
22 in any single process. If desired, this feature can be disabled by supplying
23 the "--disable-asteriskssl" option to the configure script.
25 * A new make target, 'full', has been added to the Makefile. This performs
26 the same compilation actions as make all, but will also scan the entirety of
27 each source file for documentation. This option is needed to generate AMI
28 event documentation. Note that your system must have Python in order for
29 this make target to succeed.
31 * The optimization portion of the build system has been reworked to avoid
32 broken builds on certain architectures. All architecture-specific
33 optimization has been removed in favor of using -march=native to allow gcc
34 to detect the environment in which it is running when possible. This can
35 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
37 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
38 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
40 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
41 previously parsed the header file to obtain the version of Asterisk, you
42 will now have to go through Asterisk to get the version information.
50 * Added 'F()' option. Similar to the dial option, this can be supplied with
51 arguments indicating where the callee should go after the caller is hung up,
52 or without options specified, the priority after the Queue will be used.
57 * Added menu action admin_toggle_mute_participants. This will mute / unmute
58 all non-admin participants on a conference. The confbridge configuration
59 file also allows for the default sounds played to all conference users when
60 this occurs to be overriden using sound_participants_unmuted and
61 sound_participants_muted.
63 * Added menu action participant_count. This will playback the number of
64 current participants in a conference.
66 * Added announcement configuration option to user profile. If set the sound
67 file will be played to the user, and only the user, upon joining the
73 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
74 channels respectively before the callee channels are called.
79 * Added support for IPv6.
81 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
82 external process will cause the current playlist to be cleared, including
83 stopping any audio file that is currently playing. This is useful when you
84 want to interrupt audio playback only when specific DTMF is entered by the
90 * A new option, 'I' has been added to app_followme. By setting this option,
91 Asterisk will not update the caller with connected line changes when they
92 occur. This is similar to app_dial and app_queue.
94 * The 'N' option is now ignored if the call is already answered.
96 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
97 and caller channels respectively before the callee channels are called.
99 * The winning FollowMe outgoing call is now put on hold if the caller put it on
105 * MixMonitor hooks now have IDs associated with them which can be used to
106 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
107 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
108 now accepts that ID as an argument.
110 * Added 'm' option, which stores a copy of the recording as a voicemail in the
116 * Increased the default number of allowed destinations from 5 to 12.
121 * The app_page application now no longer depends on DAHDI or app_meetme. It
122 has been re-architected to use app_confbridge internally.
127 * Added queue options autopausebusy and autopauseunavail for automatically
128 pausing a queue member when their device reports busy or congestion.
130 * The 'ignorebusy' option for queue members has been deprecated in favor of
131 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
132 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
133 per interface basis. Individual ringinuse values can now be set in
134 queues.conf via an argument to member definitions. Lastly, the queue
135 'ringinuse' setting now only determines defaults for the per member
136 'ringinuse' setting and does not override per member settings like it does
139 * Added 'F()' option. Similar to the dial option, this can be supplied with
140 arguments indicating where the callee should go after the caller is hung up,
141 or without options specified, the priority after the Queue will be used.
143 * Added new option log_member_name_as_agent, which will cause the membername to
144 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
145 state_interface has been set.
150 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
151 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
152 changed arguments to SayUnixTime so that every option is truly optional even
153 when using multiple options (so that j option could be used without having to
154 manually specify timezone and format) There are other benefits, e.g., format
155 can now be used without specifying time zone as well.
160 * Addition of the VM_INFO function - see Function changes.
162 * The imapserver, imapport, and imapflags configuration options can now be
163 overriden on a user by user basis.
165 * When voicemail plays a message's envelope with saycid set to yes, when
166 reaching the caller id field it will play a recording of a file with the same
167 base name as the sender's callerid if there is a similarly named file in
168 <astspooldir>/recordings/callerids/
170 * Voicemails now contains a unique message identifier "msg_id", which is stored
171 in the message envelope with the sound files. IMAP backends will now store
172 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
173 backends will store the message identifier in a "msg_id" column. See
174 UPGRADE.txt for more information.
176 * Added VoiceMailPlayMsg application. This application will play a single
177 voicemail message from a mailbox. The result of the application, SUCCESS or
178 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
183 * Hangup handlers can be attached to channels using the CHANNEL() function.
184 Hangup handlers will run when the channel is hung up similar to the h
185 extension. The hangup_handler_push option will push a GoSub compatible
186 location in the dialplan onto the channel's hangup handler stack. The
187 hangup_handler_pop option will remove the last added location, and optionally
188 replace it with a new GoSub compatible location. The hangup_handler_wipe
189 option will remove all locations on the stack, and optionally add a new
192 * The expression parser now recognizes the ABS() absolute value function,
193 which will convert negative floating point values to positive values.
195 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
196 control of faxdetect.
198 * Addition of the VM_INFO function that can be used to retrieve voicemail
199 user information, such as the email address and full name.
200 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
203 * The REDIRECTING function now supports the redirecting original party id
206 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
207 lets you set some of the configuration options from the [general] section
208 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
209 the key sequence used to activate built-in features, such as blindxfer,
210 and automon. See the built-in documentation for details.
212 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
213 instead of simply the uri. This is the format that MessageSend() can use
214 in the from parameter for outgoing SIP messages.
216 * Added the PRESENCE_STATE function. This allows retrieving presence state
217 information from any presence state provider. It also allows setting
218 presence state information from a CustomPresence presence state provider.
219 See AMI/CLI changes for related commands.
227 * Added a manager event "LocalBridge" for local channel call bridges between
228 the two pseudo-channels created.
233 * Added dialtone_detect option for analog ports to disconnect incoming
234 calls when dialtone is detected.
236 * Added option colp_send to send ISDN connected line information. Allowed
237 settings are block, to not send any connected line information; connect, to
238 send connected line information on initial connect; and update, to send
239 information on any update during a call. Default is update.
244 * A new channel driver named chan_motif has been added which provides support for
245 Google Talk and Jingle in a single channel driver. This new channel driver includes
246 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
247 hold, unhold, and ringing notification. It is also compliant with the current Jingle
248 specification, current Google Jingle specification, and the original Google Talk
254 * Added NAT support for RTP. Setting in config is 'nat', which can be set
255 globally and overriden on a peer by peer basis.
257 * Direct media functionality has been added. Options in config are:
258 directmedia (directrtp) and directrtpsetup (earlydirect)
260 * ChannelUpdate events now contain a CallRef header.
265 * Asterisk will no longer substitute CID number for CID name in the display
266 name field if CID number exists without a CID name. This change improves
267 compatibility with certain device features such as Avaya IP500's directory
270 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
271 created using that setting to not be removed during SIP reload.
273 * Added settings recordonfeature and recordofffeature. When receiving an INFO
274 request with a "Record:" header, this will turn the requested feature on/off.
275 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
276 dynamic features must be enabled and configured properly on the requesting
277 channel for this to function properly.
279 * Add support to realtime for the 'callbackextension' option.
281 * When multiple peers exist with the same address, but differing
282 callbackextension options, incoming requests that are matched by address
283 will be matched to the peer with the matching callbackextension if it is
286 * Two new NAT options, auto_force_rport and auto_comedia, have been added
287 which set the force_rport and comedia options automatically if Asterisk
288 detects that an incoming SIP request crossed a NAT after being sent by
291 * NAT settings are now a combinable list of options. The equivalent of the
292 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
294 * Adds an option send_diversion which can be disabled to prevent
295 diversion headers from automatically being added to INVITE requests.
297 * Add support for lightweight NAT keepalive. If enabled a blank packet will
298 be sent to the remote host at a given interval to keep the NAT mapping open.
299 This can be enabled using the keepalive configuration option.
301 * Add option 'tonezone' to specify country code for indications. This option
302 can be set both globally and overridden for specific peers.
304 * The SIP Security Events Framework now supports IPv6.
306 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
307 between multiple user agents. When set, for directmedia reinvites,
308 Asterisk will not send an immediate reinvite on an incoming call leg. This
309 option is useful when peered with another SIP user agent that is known to
310 send immediate direct media reinvites upon call establishment.
312 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
314 * Add options subminexpiry and submaxexpiry to set limits of subscription
315 timer independently from registration timer settings. The setting of the
316 registration timer limits still is done by options minexpiry, maxexpiry
317 and defaultexpiry. For backwards compatibility the setting of minexpiry
318 and maxexpiry also is used to configure the subscription timer limits if
319 subminexpiry and submaxexpiry are not set in sip.conf.
320 * Set registration timer limits to default values when reloading sip
321 configuration and values are not set by configuration.
323 * When a MESSAGE request is received, the address the request was received from
324 is now saved in the SIP_RECVADDR variable.
326 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
327 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
328 the ANI2/OLI information is set on the channel, which can be retrieved using
329 the CALLERID function.
331 * Peers can now be configured to support negotiation of ICE candidates using
332 the setting icesupport. See res_rtp_asterisk changes for more information.
334 * Added support for format attribute negotiation. See the Codecs changes for
340 * Added skinny version 17 protocol support.
345 * Added ability to use multiple lines for a single phone. This allows multiple
346 calls to occur on a single phone, using callwaiting and switching between calls.
348 * Added option 'sharpdial' allowing end dialing by pressing # key
350 * Added option 'interdigit_timer' to control phone dial timeout
352 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
354 * Added global 'debug' option, that enables debug in channel driver
356 * Added ability to translate on-screen menu in multiple languages. Tested on
357 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
358 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
361 * In addition to English added French and Russian languages for on-screen menus
363 * Reworked dialing number input: added dialing by timeout, immediate dial on
364 on dialplan compare, phone number length now not limited by screen size
366 * Added ability to pickup a call using features.conf defined value and
372 * The minimum DTMF duration can now be configured in asterisk.conf
373 as "mindtmfduration". The default value is (as before) set to 80 ms.
374 (previously it was only available in source code)
376 * Named ACLs can now be specified in acl.conf and used in configurations that
377 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
378 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
379 working ACL. In addition, some CLI commands have been added to provide
380 show information and allow for module reloading - see CLI Changes.
382 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
383 be used within the dynamic weight attribute when specifying a mapping.
385 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
386 header, instead of putting the user defined event name there. When enabled
387 the UserDefType header is added for user defined events. This feature is
388 enabled with the setting show_user_defined.
390 * Macro has been deprecated in favor of GoSub. For redirecting and connected
391 line purposes use the following variables instead of their macro equivalents:
392 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
393 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
394 cc_callback_macro in channel configurations.
399 * A new channel variable, AGIEXITONHANGUP, has been added which allows
400 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
401 AGI application would exit immediately after a channel hangup is detected.
403 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
404 are resolved and each address is attempted in turn until one succeeds or
408 AMI (Asterisk Manager Interface)
410 * Originate now generates an error response if the extension given is not found
413 * MixMonitor will now show IDs associated with the mixmonitor upon creating
414 them if the i(variable) option is used. StopMixMonitor will accept
415 MixMonitorID as an option to close specific MixMonitors.
417 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
418 updated to include information about peers configured with
419 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
420 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
421 returned if auto_force_rport is not enabled.
423 * Hangup now can take a regular expression as the Channel option. If you want
424 to hangup multiple channels, use /regex/ as the Channel option. Existing
425 behavior to hanging up a single channel is unchanged, but if you pass a regex,
426 the manager will send you a list of channels back that were hung up.
428 * Support for IPv6 addresses has been added.
430 * AMI Events can now be documented in the Asterisk source. Note that AMI event
431 documentation is only generated when Asterisk is compiled using 'make full'.
432 See the CLI section for commands to display AMI event information.
434 * The AMI Hangup event now includes the AccountCode header so you can easily
435 correlate with AMI Newchannel events.
437 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
438 the StateInterface of the queue member.
440 * Added AMI event SessionTimeout in the Call category that is issued when a
441 call is terminated due to either RTP stream inactivity or SIP session timer
444 * CEL events can now contain a user defined header UserDefType. See core
445 changes for more information.
447 * OOH323 ChannelUpdate events now contain a CallRef header.
449 * Added PresenceState command. This command will report the presence state for
450 the given presence provider.
452 * Added Parkinglots command. This will list all parking lots as a series of
453 AMI Parkinglot events.
455 * Added MessageSend command. This behaves in the same manner as the
456 MessageSend application, and is a technolgoy agnostic mechanism to send out
457 of call text messages.
459 * Added "message" class authorization. This grants an account permission to
460 send out of call messages. Write-only.
465 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
466 filenames of all running mixmonitors on a channel.
468 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
469 numeric instead of 0, 1, or 2.
471 * "stun show status" will show a table describing how the STUN client is
474 * "acl show [named acl]" will show information regarding a Named ACL. The
475 acl module can be reloaded with "reload acl".
477 * Added CLI command to display AMI event information - "manager show events",
478 which shows a list of all known and documented AMI events, and "manager show
479 event [event name]", which shows detail information about a specific AMI
482 * The result of the CLI command "queue show" now includes the state interface
483 information of the queue member.
485 * The command "core set verbose" will now set a separate level of logging for
486 each remote console without affecting any other console.
488 * Added command "cdr show pgsql status" to check connection status
490 * "sip show channel" will now display the complete route set.
492 * Added "presencestate list" command. This command will list all custom
493 presence states that have been set by using the PRESENCE_STATE dialplan
496 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
497 command. This changes a custom presence to a new state.
502 * Codec lists may now be modified by the '!' character, to allow succinct
503 specification of a list of codecs allowed and disallowed, without the
504 requirement to use two different keywords. For example, to specify all
505 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
507 * Add support for parsing SDP attributes, generating SDP attributes, and
508 passing it through. This support includes codecs such as H.263, H.264, SILK,
509 and CELT. You are able to set up a call and have attribute information pass.
510 This should help considerably with video calls.
515 * Asterisk version and build information is now logged at the beginning of a
518 * Threads belonging to a particular call are now linked with callids which get
519 added to any log messages produced by those threads. Log messages can now be
520 easily identified as involved with a certain call by looking at their call id.
521 Call ids may also be attached to log messages for just about any case where
522 it can be determined to be related to a particular call.
524 * Each logging destination and console now have an independent notion of the
525 current verbosity level. Logger.conf now allows an optional argument to
526 the 'verbose' specifier, indicating the level of verbosity sent to that
527 particular logging destination. Additionally, remote consoles now each
528 have their own verbosity level. The command 'core set verbose' will now set
529 a separate level for each remote console without affecting any other
535 * Added 'announcement' option which will play at the start of MOH and between
536 songs in modes of MOH that can detect transitions between songs (eg.
542 * New per parking lot options: comebackcontext and comebackdialtime. See
543 configs/features.conf.sample for more details.
545 * Channel variable PARKER is now set when comebacktoorigin is disabled in
548 * Channel variable PARKEDCALL is now set with the name of the parking lot
549 when a timeout occurs.
555 CDR Postgresql Driver
557 * Added command "cdr show pgsql status" to check connection status
560 CDR Adaptive ODBC Driver
562 * Added schema option for databases that support specifying a schema.
570 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
571 CALENDAR_WRITE has completed successfully.
576 * A new option, 'probation' has been added to rtp.conf
577 RTP in strictrtp mode can now require more than 1 packet to exit learning
578 mode with a new source (and by default requires 4). The probation option
579 allows the user to change the required number of packets in sequence to any
580 desired value. Use a value of 1 to essentially restore the old behavior.
581 Also, with strictrtp on, Asterisk will now drop all packets until learning
582 mode has successfully exited. These changes are based on how pjmedia handles
583 media sources and source changes.
585 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
586 enabled or disabled using the icesupport setting. A variety of other
587 settings have been introduced to configure STUN/TURN connections.
592 * A new module, res_corosync, has been introduced. This module uses the
593 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
594 of Asterisk servers to both Message Waiting Indication (MWI) and/or
595 Device State (presence) information. This module is very similar to, and
596 is a replacement for the res_ais module that was in previous releases of
602 * This module adds a cleaned up, drop-in replacement for res_jabber called
603 res_xmpp. This provides the same externally facing functionality but is
604 implemented differently internally. res_jabber has been deprecated in favor
605 of res_xmpp; please see the UPGRADE.txt file for more information.
610 * The safe_asterisk script has been updated to allow several of its parameters
611 to be set from environment variables. This also enables a custom run
612 directory of Asterisk to be specified, instead of defaulting to /tmp.
614 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
615 its value to determine the directory to assume is the top-level directory of
616 the source tree. If the variable is not set, it defaults to the current
617 behavior and uses the current working directory.
620 ------------------------------------------------------------------------------
621 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
622 ------------------------------------------------------------------------------
626 * Asterisk now has protocol independent support for processing text messages
627 outside of a call. Messages are routed through the Asterisk dialplan.
628 SIP MESSAGE and XMPP are currently supported. There are options in
629 jabber.conf and sip.conf to allow enabling these features.
630 -> jabber.conf: see the "sendtodialplan" and "context" options.
631 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
632 and "outofcall_message_context" options.
633 The MESSAGE() dialplan function and MessageSend() application have been
634 added to go along with this functionality. More detailed usage information
635 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
636 * If real-time text support (T.140) is negotiated, it will be preferred for
637 sending text via the SendText application. For example, via SIP, messages
638 that were once sent via the SIP MESSAGE request would be sent via RTP if
639 T.140 text is negotiated for a call.
643 * parkedmusicclass can now be set for non-default parking lots.
645 Asterisk Manager Interface
646 --------------------------
647 * PeerStatus now includes Address and Port.
648 * Added Hold events for when the remote party puts the call on and off hold
649 for chan_dahdi ISDN channels.
650 * Added new action MeetmeListRooms to list active conferences (shows same
651 data as "meetme list" at the CLI).
652 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
653 Description field that is set by 'description' in the channel configuration
655 * Added Uniqueid header to UserEvent.
656 * Added new action FilterAdd to control event filters for the current session.
657 This requires the system permission and uses the same filter syntax as
658 filters that can be defined in manager.conf
659 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
660 versions had some instances of the event converted, but others were left
661 as-is. All Unlink events should now be converted to Bridge events. The AMI
662 protocol version number was incremented to 1.2 as a result of this change.
665 --------------------------
666 * The HTTP Server can bind to IPv6 addresses.
669 --------------------------
670 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
671 with busydetect. usage example: busypattern=200,200,200,600
674 --------------------------
675 * New 'gtalk show settings' command showing the current settings loaded from
677 * The 'logger reload' command now supports an optional argument, specifying an
678 alternate configuration file to use.
679 * 'dialplan add extension' command will now automatically create a context if
680 the specified context does not exist with a message indicated it did so.
681 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
682 Description field which can be populated with 'description' in the channel
683 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
686 --------------------------
687 * The filter option in cdr_adaptive_odbc now supports negating the argument,
688 thus allowing records which do NOT match the specified filter.
689 * Added ability to log CONGESTION calls to CDR
692 --------------------------
693 * Ability to define custom SILK formats in codecs.conf.
694 * Addition of speex32 audio format with translation.
695 * CELT codec pass-through support and ability to define
696 custom CELT formats in codecs.conf.
697 * Ability to read raw signed linear files with sample rates
698 ranging from 8khz - 192khz. The new file extensions introduced
699 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
700 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
701 Skinny, H.323, etc) can still only support the following codecs:
702 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
703 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
704 Video: h261, h263, h263p, h264, mpeg4
709 --------------------------
710 * New highly optimized and customizable ConfBridge application capable of
711 mixing audio at sample rates ranging from 8khz-96khz.
712 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
713 and bridge profiles on a channel.
714 * CONFBRIDGE_INFO dialplan function capable of retrieving information
715 about a conference such as locked status and number of parties, admins,
717 * Addition of video_mode option in confbridge.conf for adding video support
718 into a bridge profile.
719 * Addition of the follow_talker video_mode in confbridge.conf. This video
720 mode dynamically switches the video feed to always display the loudest talker
721 supplying video in the conference.
725 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
726 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
727 variables from asterisk.conf.
731 * Addition of the JITTERBUFFER dialplan function. This function allows
732 for jitterbuffering to occur on the read side of a channel. By using
733 this function conference applications such as ConfBridge and MeetMe can
734 have the rx streams jitterbuffered before conference mixing occurs.
735 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
737 * Added STRREPLACE function. This function let's the user search a variable
738 for a given string to replace with another string as many times as the
739 user specifies or just throughout the whole string.
740 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
741 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
742 * Added extensions to chan_ooh323 in function CHANNEL()
744 libpri channel driver (chan_dahdi) DAHDI changes
745 --------------------------
746 * Added moh_signaling option to specify what to do when the channel's bridged
747 peer puts the ISDN channel on hold.
748 * Added display_send and display_receive options to control how the display ie
749 is handled. To send display text from the dialplan use the SendText()
750 application when the option is enabled.
751 * Added mcid_send option to allow sending a MCID request on a span.
754 --------------------------
755 * Added setvar option to calendar.conf to allow setting channel variables on
756 notification channels.
757 * Added "calendar show types" CLI command to list registered calendar
761 --------------------------
762 * Added two new options, r and t with file name arguments to record
763 single direction (unmixed) audio recording separate from the bidirectional
764 (mixed) recording. The mixed file name argument is optional now as long
765 as at least one recording option is used.
768 --------------------------
769 * Added a new option, l, which will disable local call optimization for
770 channels involved with the FollowMe thread. Use this option to improve
771 compatability for a FollowMe call with certain dialplan apps, options, and
775 --------------------------
776 * Added option "k" that will automatically close the conference when there's
777 only one person left when a user exits the conference.
780 --------------------------
781 * cel_pgsql now supports the 'extra' column for data added using the
782 CELGenUserEvent() application.
785 --------------------------
786 * Support for defining hints has been added to pbx_lua. See the 'hints' table
787 in the sample extensions.lua file for syntax details.
788 * Applications that perform jumps in the dialplan such as Goto will now
789 execute properly. When pbx_lua detects that the context, extension, or
790 priority we are executing on has changed it will immediately return control
791 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
792 the priority after the currently executing priority.
793 * An autoservice is now started by default for pbx_lua channels. It can be
794 stopped and restarted using the autoservice_stop() and autoservice_start()
798 --------------------------
799 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
800 into a FAXStatus event with an 'Operation' header that will be either
801 'send', 'receive', and 'gateway'.
802 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
803 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
804 feature will handle converting a fax call between an audio T.30 fax terminal
805 and an IFP T.38 fax terminal.
809 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
810 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
811 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
815 * Added general option negative_penalty_invalid default off. when set
816 members are seen as invalid/logged out when there penalty is negative.
817 for realtime members when set remove from queue will set penalty to -1.
818 * Added queue option autopausedelay when autopause is enabled it will be
819 delayed for this number of seconds since last successful call if there
820 was no prior call the agent will be autopaused immediately.
821 * Added member option ignorebusy this when set and ringinuse is not
822 will allow per member control of multiple calls as ringinuse does for
824 * Added global option check_state_unknown to enforce checking of device state
825 when the device state is unknown app_queue will see unknown as available.
829 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
831 * Added 'k' option to MeetMe to automatically kill the conference when there's only
832 one participant left (much like a normal call bridge)
833 * Added extra argument to Originate to set timeout.
837 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
838 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
839 utility in the UTILS section of menuselect. If an existing astdb is found and no
840 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
841 convert an existing astdb to the SQLite3 version automatically at runtime.
845 * Modules marked as deprecated are no longer marked as building by default. Enabling
846 these modules is still available via menuselect.
850 * authdebug is now disabled by default. To enable this functionaility again
851 set authdebug = yes in iax.conf.
855 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
856 releases it was disabled.
860 * The PBX core previously made a call with a non-existing extension test for
861 extension s@default and jump there if the extension existed.
862 This was a bad default behaviour and violated the principle of least surprise.
863 It has therefore been changed in this release. It may affect some
864 applications and configurations that rely on this behaviour. Most channel
865 drivers have avoided this for many releases by testing whether the extension
866 called exists before starting the PBX and generating a local error.
867 This behaviour still exists and works as before.
869 Extension "s" is used when no extension is given in a channel driver,
870 like immediate answer in DAHDI or calling to a domain with no user part
873 ------------------------------------------------------------------------------
874 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
875 ------------------------------------------------------------------------------
879 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
880 now defaults to force_rport. It is very important that phones requiring nat=no be
881 specifically set as such instead of relying on the default setting. If at all
882 possible, all devices should have nat settings configured in the general section as
883 opposed to configuring nat per-device.
884 * Added preferred_codec_only option in sip.conf. This feature limits the joint
885 codecs sent in response to an INVITE to the single most preferred codec.
886 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
887 to be used for the outgoing call. It must be one of the codecs configured
889 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
890 to be used for holding a private key. If tlsprivatekey is not specified,
891 tlscertfile is searched for both public and private key.
892 * Added tlsclientmethod option to sip.conf. This allows the protocol for
893 outbound client connections to be specified.
894 * The sendrpid parameter has been expanded to include the options
895 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
896 header to be sent (equivalent to setting sendrpid=yes) and setting
897 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
898 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
899 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
900 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
901 will accept the SDP even if the SDP version number is not properly incremented,
902 but will generate a warning in the log indicating that the SIP peer that sent
903 the SDP should have the 'ignoresdpversion' option set.
904 * The 'nat' option has now been been changed to have yes, no, force_rport, and
905 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
906 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
907 remote side requests it and disables symmetric RTP support. Setting it to
908 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
909 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
910 and enables symmetric RTP support.
911 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
912 response. This permits the master channel to know how each channel dialled
913 in a multi-channel setup resolved in an individual way. This carries a
914 performance penalty and can be disabled in sip.conf using the
915 'storesipcause' option.
916 * Added 'externtcpport' and 'externtlsport' options to allow custom port
917 configuration for the externip and externhost options when tcp or tls is used.
918 * Added support for message body (stored in content variable) to SIP NOTIFY message
919 accessible via AMI and CLI.
920 * Added 'media_address' configuration option which can be used to explicitly specify
921 the IP address to use in the SDP for media (audio, video, and text) streams.
922 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
923 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
925 * Added 'use_q850_reason' configuration option for generating and parsing
926 if available Reason: Q.850;cause=<cause code> header. It is implemented
927 in some gateways for better passing PRI/SS7 cause codes via SIP.
928 * When dialing SIP peers, a new component may be added to the end of the dialstring
929 to indicate that a specific remote IP address or host should be used when dialing
930 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
931 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
932 ability to selectively force bridged channels to also be encrypted is also
933 implemented. Branching in the dialplan can be done based on whether or not
934 a channel has secure media and/or signaling.
935 * Added directmediapermit/directmediadeny to limit which peers can send direct media
937 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
938 Charge messages to snom phones.
939 * Added support for G.719 media streams.
940 * Added support for 16khz signed linear media streams.
941 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
942 RTP has been outfitted with the same abilities.
943 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
944 available in device configurations as well as in the dial plan.
945 * Addition of the 'subscribe_network_change' option for turning on and off
946 res_stun_monitor module support in chan_sip.
947 * Addition of the 'auth_options_requests' option for turning on and off
948 authentication for OPTIONS requests in chan_sip.
952 * Add #tryinclude statement for config files. This provides the same
953 functionality as the #include statement however an asterisk module will
954 still load if the filename does not exist. Using the #include statement
955 Asterisk will not allow the module to load.
959 * Added rtsavesysname option into iax.conf to allow the systname to be saved
961 * Added the ability for chan_iax2 to inform the dialplan whether or not
962 encryption is being used. This interoperates with the SIP SRTP implementation
963 so that a secure SIP call can be bridged to a secure IAX call when the
964 dialplan requires bridged channels to be "secure".
965 * Addition of the 'subscribe_network_change' option for turning on and off
966 res_stun_monitor module support in chan_iax.
971 * Added ability to preset channel variables on indicated lines with the setvar
972 configuration option. Also, clearvars=all resets the list of variables back
974 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
975 See configs/res_pktccops.conf for more information.
977 XMPP Google Talk/Jingle changes
978 -------------------------------
979 * Added the externip option to gtalk.conf.
980 * Added the stunaddr option to gtalk.conf which allows for the automatic
981 retrieval of the external ip from a stun server.
985 * Added 'p' option to PickupChan() to allow for picking up channel by the first
986 match to a partial channel name.
987 * Added .m3u support for Mp3Player application.
988 * Added progress option to the app_dial D() option. When progress DTMF is
989 present, those values are sent immediately upon receiving a PROGRESS message
990 regardless if the call has been answered or not.
991 * Added functionality to the app_dial F() option to continue with execution
992 at the current location when no parameters are provided.
993 * Added the 'a' option to app_dial to answer the calling channel before any
994 announcements or macros are executed.
995 * Modified app_dial to set answertime when the called channel answers even if
996 the called channel hangs up during playback of an announcement.
997 * Modified app_dial 'r' option to support an additional parameter to play an
998 indication tone from indications.conf
999 * Added c() option to app_chanspy. This option allows custom DTMF to be set
1000 to cycle through the next available channel. By default this is still '*'.
1001 * Added x() option to app_chanspy. This option allows DTMF to be set to
1002 exit the application.
1003 * The Voicemail application has been improved to automatically ignore messages
1004 that only contain silence.
1005 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
1006 associated mailbox(es) to be greetings-only.
1007 * The ChanSpy application now has the 'S' option, which makes the application
1008 automatically exit once it hits a point where no more channels are available
1010 * The ChanSpy application also now has the 'E' option, which spies on a single
1011 channel and exits when that channel hangs up.
1012 * The MeetMe application now turns on the DENOISE() function by default, for
1013 each participant. In our tests, this has significantly decreased background
1014 noise (especially noisy data centers).
1015 * Voicemail now permits storage of secrets in a separate file, located in the
1016 spool directory of each individual user. The control for this is located in
1017 the "passwordlocation" option in voicemail.conf. Please see the sample
1018 configuration for more information.
1019 * The ChanIsAvail application now exposes the returned cause code using a separate
1020 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
1021 * Added 'd' option to app_followme. This option disables the "Please hold"
1023 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
1024 received will terminate recording.
1025 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
1026 Previously the folder could only be set per context, but has now been extended
1027 using the imapfolder option.
1028 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
1029 * Voicemail now allows the pager date format to be specified separately from the
1031 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
1032 to allow joining, leaving, and sending text to group chats.
1033 * MeetMe has a new option 'G' to play an announcement before joining a conference.
1034 * Page has a new option 'A(x)' which will playback an announcement simultaneously
1035 to all paged phones (and optionally excluding the caller's one using the new
1036 option 'n') before the call is bridged.
1037 * The 'f' option to Dial has been augmented to take an optional argument. If no
1038 argument is provided, the 'f' option works as it always has. If an argument is
1039 provided, then the connected party information of all outgoing channels created
1040 during the Dial will be set to the argument passed to the 'f' option.
1041 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
1043 * The OSP lookup application adds in/outbound network ID, optional security,
1044 number portability, QoS reporting, destination IP port, custom info and service
1046 * Added new application VMSayName that will play the recorded name of the voicemail
1047 user if it exists, otherwise will play the mailbox number.
1048 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
1049 retrieve state for a particular bridge, where <name> is the conference name
1050 * app_directory now allows exiting at any time using the operator or pound key.
1051 * Voicemail now supports setting a locale per-mailbox.
1052 * Two new applications are provided for declining counting phrases in multiple
1053 languages. See the application notes for SayCountedNoun and SayCountedAdj for
1055 * Voicemail now runs the externnotify script when pollmailboxes is activated and
1057 * Voicemail now includes rdnis within msgXXXX.txt file.
1058 * ExternalIVR now supports IPv6 addresses.
1059 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
1060 at https://wiki.asterisk.org/wiki/x/oQBB
1061 * ParkedCall and Park can now specify the parking lot to use.
1065 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
1066 over SRV records associated with a specific service. From the CLI, type
1067 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
1068 details on how these may be used.
1069 * PITCH_SHIFT dialplan function added. This function can be used to modify the
1070 pitch of a channel's tx and rx audio streams.
1071 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
1072 setting various connected line and redirecting party information.
1073 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
1074 support ISDN subaddressing.
1075 * The CHANNEL() function now supports the "name" and "checkhangup" options.
1076 * For DAHDI channels, the CHANNEL() dialplan function now allows
1077 the dialplan to request changes in the configuration of the active
1078 echo canceller on the channel (if any), for the current call only.
1081 exten => s,n,Set(CHANNEL(echocan_mode)=off)
1083 The possible values are:
1085 on - normal mode (the echo canceller is actually reinitialized)
1087 fax - FAX/data mode (NLP disabled if possible, otherwise completely
1089 voice - voice mode (returns from FAX mode, reverting the changes that
1090 were made when FAX mode was requested)
1091 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
1092 and setting variables on the channel which created the current channel.
1093 Administrators should take care to avoid naming conflicts, when multiple
1094 channels are dialled at once, especially when used with the Local channel
1095 construct (which all could set variables on the master channel). Usage
1096 of the HASH() dialplan function, with the key set to the name of the slave
1097 channel, is one approach that will avoid conflicts.
1098 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
1100 * func_odbc now allows multiple row results to be retrieved without using
1101 mode=multirow. If rowlimit is set, then additional rows may be retrieved
1102 from the same query by using the name of the function which retrieved the
1103 first row as an argument to ODBC_FETCH().
1104 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
1105 dialplan. This function returns the content of the received message.
1106 * Added REPLACE, which searches a given variable name for a set of characters,
1107 then either replaces them with a single character or deletes them.
1108 * Added PASSTHRU, which literally passes the same argument back as its return
1109 value. The intent is to be able to use a literal string argument to
1110 functions that currently require a variable name as an argument.
1111 * HASH-associated variables now can be inherited across channel creation, by
1112 prefixing the name of the hash at assignment with the appropriate number of
1113 underscores, just like variables.
1114 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
1115 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
1116 whether or not channels that are bridged to the current channel will be
1117 required to have secure signaling and/or media.
1118 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
1119 the current channel has secure signaling and/or media.
1120 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
1121 "no_media_path" option.
1122 Returns "0" if there is a B channel associated with the call.
1123 Returns "1" if no B channel is associated with the call. The call is either
1124 on hold or is a call waiting call.
1125 * Added option to dialplan function CDR(), the 'f' option
1126 allows for high resolution times for billsec and duration fields.
1127 * FILE() now supports line-mode and writing.
1128 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
1129 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
1133 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
1134 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
1135 and is set when a dynamic feature is triggered.
1136 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
1137 to dynamically create a new parking lot matching the value this varible is
1139 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
1140 features.conf that should be the base for dynamic parkinglots.
1141 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
1142 parkinglot should have.
1143 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
1144 parkinglot should have.
1145 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
1150 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
1151 timeout has expired.
1152 * Added 'R' option to app_queue. This option stops moh and indicates ringing
1153 to the caller when an Agent's phone is ringing. This can be used to indicate
1154 to the caller that their call is about to be picked up, which is nice when
1155 one has been on hold for an extened period of time.
1156 * A new config option, penaltymemberslimit, has been added to queues.conf.
1157 When set this option will disregard penalty settings when a queue has too
1159 * A new option, 'I' has been added to both app_queue and app_dial.
1160 By setting this option, Asterisk will not update the caller with
1161 connected line changes or redirecting party changes when they occur.
1162 * A 'relative-periodic-announce' option has been added to queues.conf. When
1163 enabled, this option will cause periodic announce times to be calculated
1164 from the end of announcements rather than from the beginning.
1165 * The autopause option in queues.conf can be passed a new value, "all." The
1166 result is that if a member becomes auto-paused, he will be paused in all
1167 queues for which he is a member, not just the queue that failed to reach
1169 * Added dialplan function QUEUE_EXISTS to check if a queue exists
1170 * The queue logger now allows events to optionally propagate to a file,
1171 even when realtime logging is turned on. Additionally, realtime logging
1172 supports sending the event arguments to 5 individual fields, although it
1173 will fallback to the previous data definition, if the new table layout is
1176 mISDN channel driver (chan_misdn) changes
1177 ----------------------------------------
1178 * Added display_connected parameter to misdn.conf to put a display string
1179 in the CONNECT message containing the connected name and/or number if
1180 the presentation setting permits it.
1181 * Added display_setup parameter to misdn.conf to put a display string
1182 in the SETUP message containing the caller name and/or number if the
1183 presentation setting permits it.
1184 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
1185 indicate the dialplan settings are to be obtained from the asterisk
1187 * Made misdn.conf parameter callerid accept the "name" <number> format
1188 used by the rest of the system.
1189 * Made use the nationalprefix and internationalprefix misdn.conf
1190 parameters to prefix any received number from the ISDN link if that
1191 number has the corresponding Type-Of-Number. NOTE: This includes
1192 comparing the incoming call's dialed number against the MSN list.
1193 * Added the following new parameters: unknownprefix, netspecificprefix,
1194 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
1195 received number from the ISDN link if that number has the corresponding
1197 * Added new dialplan application misdn_command which permits controlling
1198 the CCBS/CCNR functionality.
1199 * Added new dialplan function mISDN_CC which permits retrieval of various
1200 values from an active call completion record.
1201 * For PTP, you should manually send the COLR of the redirected-to party
1202 for an incomming redirected call if the incoming call could experience
1203 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
1204 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
1205 if the REDIRECTING(from-num) is not empty.
1206 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
1207 option on all of the REDIRECTING statements before dialing the
1208 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
1209 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
1210 redirecting-to presentation (COLR) when it becomes available.
1211 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
1214 thirdparty mISDN enhancements
1215 -----------------------------
1216 mISDN has been modified by Digium, Inc. to greatly expand facility message
1218 * Enhanced COLP support for call diversion and transfer.
1219 * CCBS/CCNR support.
1221 The latest modified mISDN v1.1.x based version is available at:
1222 http://svn.digium.com/svn/thirdparty/mISDN/trunk
1223 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
1225 Tagged versions of the modified mISDN code are available under:
1226 http://svn.digium.com/svn/thirdparty/mISDN/tags
1227 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
1229 libpri channel driver (chan_dahdi) DAHDI changes
1230 -------------------------------------------
1231 * The channel variable PRIREDIRECTREASON is now just a status variable
1232 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
1233 to read and alter the reason.
1234 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
1235 redirected-to party for an incomming redirected call if the incoming call
1236 could experience further redirects. Just set the
1237 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
1238 to the COLR. A call has been redirected if the REDIRECTING(count) is not
1240 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
1241 use the inhibit(i) option on all of the REDIRECTING statements before
1242 dialing the redirected-to party. You still have to set the
1243 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
1244 will update the redirecting-to presentation (COLR) when it becomes available.
1245 * Added the ability to ignore calls that are not in a Multiple Subscriber
1246 Number (MSN) list for PTMP CPE interfaces.
1247 * Added dynamic range compression support for dahdi channels. It is
1248 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
1249 * Added support for ISDN calling and called subaddress with partial support
1250 for connected line subaddress.
1251 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
1252 * Added handling of received HOLD/RETRIEVE messages and the optional ability
1253 to transfer a held call on disconnect similar to an analog phone.
1254 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
1255 Will reroute/deflect an outgoing call when receive the message.
1256 Can use the DAHDISendCallreroutingFacility to send the message for the
1258 * Added standard location to add options to chan_dahdi dialing:
1259 Dial(DAHDI/g1[/extension[/options]])
1262 R Reverse charging indication
1263 * Added Reverse Charging Indication (Collect calls) send/receive option.
1264 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
1265 Dial(DAHDI/g1/extension/R)
1266 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
1267 (requires latest LibPRI)
1268 * Added ability to send/receive keypad digits in the SETUP message.
1269 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
1270 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
1271 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
1272 (requires latest LibPRI)
1273 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
1274 to eliminate tromboned calls. A tromboned call goes out an interface and comes
1275 back into the same interface. Tromboned calls happen because of call routing,
1276 call deflection, call forwarding, and call transfer.
1277 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
1278 * Added the ability to support call waiting calls. (The SETUP has no B channel
1280 * Added Malicious Call ID (MCID) event to the AMI call event class.
1281 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
1283 Asterisk Manager Interface
1284 --------------------------
1285 * The Hangup action now accepts a Cause header which may be used to
1286 set the channel's hangup cause.
1287 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
1288 to specify a separate .pem file to hold a private key. By default sslcert
1289 is used to hold both the public and private key.
1290 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
1291 for options containing the 'tls' prefix. For example, 'sslenable' is now
1292 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
1293 across all .conf files. All affected sample.conf files have been modified to
1294 reflect this change. Previous options such as 'sslenable' still work,
1295 but options with the 'tls' prefix are preferred.
1296 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
1297 in a channel. (res_mutestream.so)
1298 * The configuration file manager.conf now supports a channelvars option, which
1299 specifies a list of channel variables to include in each channel-oriented
1301 * The redirect command now has new parameters ExtraContext, ExtraExtension,
1302 and ExtraPriority to allow redirecting the second channel to a different
1303 location than the first.
1304 * Added new event "JabberStatus" in the Jabber module to monitor buddies
1306 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
1307 in a MixMonitor recording.
1308 * The 'iax2 show peers' output is now similar to the expected output of
1310 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
1312 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
1313 AOC-E messages on a channel.
1314 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
1315 conform more closely to similar events.
1316 * Added a new eventfilter option per user to allow whitelisting and blacklisting
1318 * Added optional parkinglot variable for park command.
1319 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
1320 if CallerIDNum and CallerIDName headers are also present.
1322 Channel Event Logging
1323 ---------------------
1324 * A new interface, CEL, is introduced here. CEL logs single events, much like
1325 the AMI, but it differs from the AMI in that it logs to db backends much
1326 like CDR does; is based on the event subsystem introduced by Russell, and
1327 can share in all its benefits; allows multiple backends to operate like CDR;
1328 is specialized to event data that would be of concern to billing sytems,
1329 like CDR. Backends for logging and accounting calls have been produced,
1330 but a new CDR backend is still in development.
1334 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
1335 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
1336 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
1337 * Multiple files and formats can now be specified in cdr_custom.conf.
1338 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
1339 See configs/cdr_syslog.conf.sample for more information.
1340 * A 'sequence' field has been added to CDRs which can be combined with
1341 linkedid or uniqueid to uniquely identify a CDR.
1342 * Handling of billsec and duration field has changed. If your table definition
1343 specifies those fields as float,double or similar they will now be logged with
1344 microsecond accuracy instead of a whole integer.
1346 Calendaring for Asterisk
1347 ------------------------
1348 * A new set of modules were added supporing calendar integration with Asterisk.
1349 Dialplan functions for reading from and writing to calendars are included,
1350 as well as the ability to execute dialplan logic upon calendar event notifications.
1351 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
1352 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
1353 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
1354 2003 support does not support forms-based authentication).
1356 Call Completion Supplementary Services for Asterisk
1357 ---------------------------------------------------
1358 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
1359 DAHDI/ISDN supports call completion for the following switch types:
1360 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
1361 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
1363 Multicast RTP Support
1364 ---------------------
1365 * A new RTP engine and channel driver have been added which supports Multicast RTP.
1366 The channel driver can be used with the Page application to perform multicast RTP
1367 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
1368 Type can be either basic or linksys.
1369 Destination is the IP address and port for the RTP packets.
1370 Control address is specific to the linksys type and is used for sending the control
1371 packets unique to them.
1373 Security Events Framework
1374 -------------------------
1375 * Asterisk has a new C API for reporting security events. The module res_security_log
1376 sends these events to the "security" logger level. Currently, AMI is the only
1377 Asterisk component that reports security events. However, SIP support will be
1378 coming soon. For more information on the security events framework, see the
1379 "Asterisk Security Framework" section of the Asterisk wiki at
1380 https://wiki.asterisk.org/wiki/x/wgBQ
1381 * SIP support was added in Asterisk 10
1382 * This API now supports IPv6 addresses
1386 * A technology independent fax frontend (res_fax) has been added to Asterisk.
1387 * A spandsp based fax backend (res_fax_spandsp) has been added.
1388 * The app_fax module has been deprecated in favor of the res_fax module and
1389 the new res_fax_spandsp backend.
1390 * The SendFAX and ReceiveFAX applications now send their log messages to a
1391 'fax' logger level, instead of to the generic logger levels. To see these
1392 messages, the system's logger.conf file will need to direct the 'fax' logger
1393 level to one or more destinations; the logger.conf.sample file includes an
1394 example of how to do this. Note that if the 'fax' logger level is *not*
1395 directed to at least one destination, log messages generated by these
1396 applications will be lost, and that if the 'fax' logger level is directed to
1397 the console, the 'core set verbose' and 'core set debug' CLI commands will
1398 have no effect on whether the messages appear on the console or not.
1402 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
1403 Now, in order to enable transmitting silence during record the transmit_silence
1404 option should be used. transmit_silence_during_record remains a valid option, but
1405 defaults to the behavior of the transmit_silence option.
1406 * Addition of the Unit Test Framework API for managing registration and execution
1407 of unit tests with the purpose of verifying the operation of C functions.
1408 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
1409 XMPP text messages to the remote JID.
1410 * Modules.conf has a new option - "require" - that marks a module as critical for
1411 the execution of Asterisk.
1412 If one of the required modules fail to load, Asterisk will exit with a return
1414 * An 'X' option has been added to the asterisk application which enables #exec support.
1415 This allows #exec to be used in asterisk.conf.
1416 * jabber.conf supports a new option auth_policy that toggles auto user registration.
1417 * A new lockconfdir option has been added to asterisk.conf to protect the
1418 configuration directory (/etc/asterisk by default) during reloads.
1419 * The parkeddynamic option has been added to features.conf to enable the creation
1420 of dynamic parkinglots.
1421 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
1422 the reportalarms config option.
1423 * chan_dahdi supports dialing configuring and dialing by device file name.
1424 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
1425 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
1426 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
1427 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
1428 Handy for the above name-based syntax as it does not depend on
1429 initialization order.
1430 * The Realtime dialplan switch now caches entries for 1 second. This provides a
1431 significant increase in performance (about 3X) for installations using this switchtype.
1432 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
1433 AIS. For more information, please see the Distributed Device State section of the
1434 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1435 * The addition of G.719 pass-through support.
1436 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
1437 during device configuration.
1438 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
1439 have less than 3 lines on the LCD.
1440 * Realtime now supports database failover. See the sample extconfig.conf for details.
1441 * The addition of improved translation path building for wideband codecs. Sample
1442 rate changes during translation are now avoided unless absolutely necessary.
1443 * The addition of the res_stun_monitor module for monitoring and reacting to network
1444 changes while behind a NAT.
1448 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
1449 optionally accept a filename, to apply the setting only to the code generated from
1450 that source file when Asterisk was built. However, there are some modules in Asterisk
1451 that are composed of multiple source files, so this did not result in the behavior
1452 that users expected. In this version, 'core set debug' and 'core set verbose'
1453 can optionally accept *module* names instead (with or without the .so extension),
1454 which applies the setting to the entire module specified, regardless of which source
1455 files it was built from.
1456 * New 'manager show settings' command showing the current settings loaded from
1458 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
1459 the channel hangup request to all channels.
1460 * Added a "core reload" CLI command that executes a global reload of Asterisk.
1462 ------------------------------------------------------------------------------
1463 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
1464 ------------------------------------------------------------------------------
1468 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
1469 Snom phones use this for call pickup of extensions that the phone is
1471 * Added support for setting the domain in the URI for caller of an
1472 outbound call by using the SIPFROMDOMAIN channel variable.
1473 * Added a new configuration option "remotesecret" for authentication to
1474 remote services. For backwards compatibility, "secret" still has the
1475 same function as before, but now you can configure both a remote secret and a
1476 local secret for mutual authentication.
1477 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
1478 the sound will be played to the target of an attended transfer
1479 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
1480 finer control over how many peers Asterisk will qualify and the gap between them
1481 when all peers need to be qualified at the same time.
1482 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
1483 (either globally or for a specific peer), chan_sip will treat any SDP data
1484 it receives as new data and update the media stream accordingly. By
1485 default, Asterisk will only modify the media stream if the SDP session
1486 version received is different from the current SDP session version. This
1487 option is required to interoperate with devices that have non-standard SDP
1488 session version implementations (observed with Microsoft OCS). This option
1489 is disabled by default.
1490 * The parsing of register => lines in sip.conf has been modified to allow a port
1491 to be present in the "user" portion. Please see the sip.conf.sample file for more
1493 * Added support for subscribing to MWI on a remote server and making the status available
1494 as a mailbox. Please see the sip.conf.sample file for more information.
1495 * Added a function to remove SIP headers added in the dialplan before the
1496 first INVITE is generated - SIPRemoveHeader()
1497 * Channel variables set with setvar= in a device configuration is now
1498 set both for inbound and outbound calls.
1499 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
1503 * Added immediate option to iax.conf
1504 * Added forceencryption option to iax.conf
1505 * Added Encryption and Trunk status to manager command "iaxpeers"
1509 * The configuration file now holds separate sections for devices and lines.
1510 Please have a look at configs/skinny.conf.sample and change your skinny.conf
1515 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
1516 support for LibOpenR2. http://www.libopenr2.org/
1517 * The UK option waitfordialtone has been added for use with BT analog
1519 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
1520 is used in conjunction with the 'faxdetect' configuration option. When
1521 'faxbuffers' is used and fax tones are detected, the channel will dynamically
1522 switch to the configured faxbuffers policy. For example, to use 6 buffers
1523 and a 'full' buffer policy for a fax transmission, add:
1525 The faxbuffers configuration will be in affect until the call is torn down.
1526 * Added service message support for 4ESS/5ESS switches.
1530 * For DAHDI channels, the CHANNEL() dialplan function now
1531 supports changing the channel's buffer policy (for the current
1532 call only), using this syntax:
1534 exten => s,n,Set(CHANNEL(buffers)=6,full)
1536 This would change the channel to the 'full' buffer policy and
1537 6 (six) buffers. Possible options for this setting are the same
1538 as those in chan_dahdi.conf.
1539 * Added a new dialplan function, CURLOPT, which permits setting various
1540 options that may be useful with the CURL dialplan function, such as
1541 cookies, proxies, connection timeouts, passwords, etc.
1542 * Permit the syntax and synopsis fields of the corresponding dialplan
1543 functions to be individually set from func_odbc.conf.
1544 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
1545 * func_odbc now may specify an insert query to execute, when the write query
1546 affects 0 rows (usually indicating that no such row exists).
1547 * Added a new dialplan function, LISTFILTER, which permits removing elements
1548 from a set list, by name. Uses the same general syntax as the existing CUT
1549 and FIELDQTY dialplan functions, which also manage lists.
1550 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
1551 obtaining realtime data from the dialplan.
1552 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
1553 a subroutine when using the GoSub() and Return() applications.
1554 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
1555 of "core show function AUDIOHOOK_INHERIT" from the CLI
1556 * Added AES_ENCRYPT. For information on its use, please see the output
1557 of "core show function AES_ENCRYPT" from the CLI
1558 * Added AES_DECRYPT. For information on its use, please see the output
1559 of "core show function AES_DECRYPT" from the CLI
1560 * func_odbc now supports database transactions across multiple queries.
1564 * Scheduled meetme conferences may now have their end times extended by
1566 * app_authenticate now gives the ability to select a prompt other than
1568 * app_directory now pays attention to the searchcontexts setting in
1569 voicemail.conf and will look through all contexts, if no context is
1570 specified in the initial argument.
1571 * A new application, Originate, has been introduced, that allows asynchronous
1572 call origination from the dialplan.
1573 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
1574 in addition to the setting in the "general" context.
1575 * Added ConfBridge dialplan application which does conference bridges without
1576 DAHDI. For information on its use, please see the output of
1577 "core show application ConfBridge" from the CLI.
1581 * The Asterisk CLI has a new command, "channel redirect", which is similar in
1582 operation to the AMI Redirect action.
1583 * extensions.conf now allows you to use keyword "same" to define an extension
1584 without actually specifying an extension. It uses exactly the same pattern
1585 as previously used on the last "exten" line. For example:
1586 exten => 123,1,NoOp(something)
1587 same => n,SomethingElse()
1588 * musiconhold.conf classes of type 'files' can now use relative directory paths,
1589 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
1590 * All deprecated CLI commands are removed from the sourcecode. They are now handled
1591 by the new clialiases module. See cli_aliases.conf.sample file.
1592 * Times within timespecs are now accurate down to the minute. This is a change
1593 from historical Asterisk, which only provided timespecs rounded to the nearest
1594 even (read: evenly divisible by 2) minute mark.
1595 * The realtime switch now supports an option flag, 'p', which disables searches for
1597 * In addition to a time range and date range, timespecs now accept a 5th optional
1598 argument, timezone. This allows you to perform time checks on alternate
1599 timezones, especially if those daylight savings time ranges vary from your
1600 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
1602 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
1603 give you the correct output for an asterisk box behind nat. It will give you the
1604 externhost and localnet settings.
1605 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
1606 can connect calls in passthrough mode, as well as record and play back files.
1607 * Successful and unsuccessful call pickup can now be alerted through sounds, by
1608 using pickupsound and pickupfailsound in features.conf.
1609 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
1610 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
1611 instead of the /var/run/asterisk.pid where it used to be. This will make
1612 installs as non-root easier to manage.
1617 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
1618 be written; they will no longer be explicitly written.
1620 Asterisk Manager Interface
1621 --------------------------
1622 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
1623 a non-empty value) in your request. If you do this, any pending AMI events will
1624 *not* be included in the response to your request as they would normally, but
1625 will be left in the event queue for the next request you make to retrieve. For
1626 some applications, this will allow you to guarantee that you will only see
1627 events in responses to 'WaitEvent' actions, and can better know when to expect them.
1628 To know whether the Asterisk server supports this header or not, your client can
1629 inspect the first response back from the server to see if it includes this header:
1631 Pragma: SuppressEvents
1633 If this is included, the server supports event suppression.
1635 * Added 4 new Actions to list skinny device(s) and line(s)
1641 LDAP Schema File Additions
1642 --------------------------
1643 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
1644 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
1646 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
1647 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
1648 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
1649 * Removed redundant IPaddr (there's already IPAddress)
1650 - Gives more configuration Flags for SIP-Users available (tested)
1651 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
1652 without extensibleObject (which really should be the last resort); gives
1653 also additional possibilities for LDAP-filter
1655 ------------------------------------------------------------------------------
1656 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
1657 ------------------------------------------------------------------------------
1659 Device State Handling
1660 ---------------------
1661 * The event infrastructure in Asterisk got another big update to help support
1662 distributed events. It currently supports distributed device state and
1663 distributed Voicemail MWI (Message Waiting Indication). A new module has
1664 been merged, res_ais, which facilitates communicating events between servers.
1665 It uses the SAForum AIS (Service Availability Forum Application Interface
1666 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
1667 a cluster of Asterisk servers, and to share events between them. For more
1668 information on setting this up, refer to the Distributed Device State section
1669 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1673 * Added a new dialplan function, AST_CONFIG(), which allows you to access
1674 variables from an Asterisk configuration file.
1675 * The JACK_HOOK function now has a c() option to supply a custom client name.
1676 * Added two new dialplan functions from libspeex for audio gain control and
1677 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
1678 rx directions of a channel from the dialplan.
1679 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
1680 based on other parameters. The default is still to search based on the
1681 forwarding station ID. However, there are new options that allow you to search
1682 based on the message desk terminal ID, or the message desk number.
1683 * TIMEOUT() has been modified to be accurate down to the millisecond.
1684 * ENUM*() functions now include the following new options:
1685 - 'u' returns the full URI and does not strip off the URI-scheme.
1686 - 's' triggers ISN specific rewriting
1687 - 'i' looks for branches into an Infrastructure ENUM tree
1688 - 'd' for a direct DNS lookup without any flipping of digits.
1689 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
1690 * CHANNEL() now has options for the maximum, minimum, and standard or normal
1691 deviation of jitter, rtt, and loss for a call using chan_sip.
1693 DAHDI channel driver (chan_dahdi) Changes
1694 ----------------------------------------
1695 * Channels can now be configured using named sections in chan_dahdi.conf, just
1696 like other channel drivers, including the use of templates.
1697 * The default for pridialplan has changed from 'national' to 'unknown'.
1701 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
1702 to something that matches the pattern a hint will be created using the contents
1703 and variables evaluated.
1704 * Dialplan matching has been extended to allow an extension to return to the
1705 PBX core to wait for more digits. This is done by using the new dialplan
1706 application called "Incomplete". This will permit a whole new level of
1707 extension control, by giving the administrator more control over early
1708 matches employing one of the short-circuit pattern match operators. Note
1709 that custom applications can trigger this same behavior by returning the
1710 special value AST_PBX_INCOMPLETE.
1714 * Directory now permits both first and last names to be matched at the same
1715 time. In addition, the number of digits to enter of the name can be set in
1716 the arguments to Directory; previously, you could enter only 3, regardless
1717 of how many names are in your company. For large companies, this should be
1719 * Voicemail now permits a mailbox setting to wrap around from first to last
1720 messages, if the "messagewrap" option is set to a true value.
1721 * Voicemail now permits an external script to be run, for password validation.
1722 The script should output "VALID" or "INVALID" on stdout, depending upon the
1723 wish to validate or invalidate the password given. Arguments are:
1724 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
1726 * Dial has a new option: F(context^extension^pri), which permits a callee to
1727 continue in the dialplan, at the specified label, if the caller hangs up.
1728 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
1729 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
1730 * The Jack application now has a c() option to supply a custom client name.
1731 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
1732 like the pre-existing whisper mode, except that the spy can also talk to the
1733 participant on the bridged channel as well.
1734 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
1735 to be spoken instead of the channel name or number. For more information on the
1736 use of this option, issue the command "core show application ChanSpy" from the
1738 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
1739 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
1740 words, if using the 'd' option, it is not possible to enter a number to append to
1741 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
1742 change to whisper mode, and pressing 6 will change to barge mode.
1743 * ExternalIVR now takes several options that affect the way it performs, as
1744 well as having several new commands. Please see the External IVR page on the Asterisk
1745 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
1746 * Added ability to communicate over a TCP socket instead of forking a child process for the
1747 ExternalIVR application.
1748 * ChanIsAvail has a new option, 'a', which will return all available channels instead
1749 of just the first one if you give the function more then one channel to check.
1750 * PrivacyManager now takes an option where you can specify a context where the
1751 given number will be matched. This way you have more control over who is allowed
1752 and it stops the people who blindly enter 10 digits.
1753 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
1754 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
1755 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
1756 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
1757 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
1758 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
1759 * The Dial() application no longer copies the language used by the caller to the callee's
1760 channel. If you desire for the caller's channel's language to be used for file playback
1761 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
1762 * SendImage() no longer hangs up the channel on error; instead, it sets the
1763 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
1764 'UNSUPPORTED'. This change makes SendImage() more consistent with other
1766 * Park has a new option, 's', which silences the announcement of the parking space number.
1767 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
1768 invalid input and will be assumed to mean that no timeout is desired.
1772 * Added DNS manager support to registrations for peers referencing peer entries.
1773 DNS manager runs in the background which allows DNS lookups to be run asynchronously
1774 as well as periodically updating the IP address. These properties allow for
1775 better performance as well as recovery in the event of an IP change.
1776 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
1777 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
1778 These changes also provide performance improvements for call setup and tear down.
1779 * Added ability to specify registration expiry time on a per registration basis in
1781 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
1783 * Added t38pt_usertpsource option. See sip.conf.sample for details.
1784 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
1785 * 'sip show peers' and 'sip show users' display their entries sorted in
1786 alphabetical order, as opposed to the order they were in, in the config
1788 * Videosupport now supports an additional option, "always", which always sets
1789 up video RTP ports, even on clients that don't support it. This helps with
1790 callfiles and certain transfers to ensure that if two video phones are
1791 connected, they will always share video feeds.
1795 * Existing DNS manager lookups extended to check for SRV records.
1796 * IAX2 encryption support has been improved to support periodic key rotation
1797 within a call for enhanced security. The option "keyrotate" has been
1798 provided to disable this functionality to preserve backwards compatibility
1799 with older versions of IAX2 that do not support key rotation.
1803 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
1804 data tree based on the given <path>.
1805 * New CLI command "data show providers" that will display all the registered
1807 * New CLI command, "config reload <file.conf>" which reloads any module that
1808 references that particular configuration file. Also added "config list"
1809 which shows which configuration files are in use.
1810 * New CLI commands, "pri show version" and "ss7 show version" that will
1811 display which version of libpri and libss7 are being used, respectively.
1812 A new API call was added so trunk will now have to be compiled against
1813 a versions of libpri and libss7 that have them or it will not know that
1814 these libraries exist.
1815 * The commands "core show globals", "core set global" and "core set chanvar" has
1816 been deprecated in favor of the more semanticly correct "dialplan show globals",
1817 "dialplan set chanvar" and "dialplan set global".
1818 * New CLI command "dialplan show chanvar" to list all variables associated
1819 with a given channel.
1823 * Addresses managed by DNS manager now can check to see if there is a DNS
1824 SRV record for a given domain and will use that hostname/port if present.
1826 AMI - The manager (TCP/TLS/HTTP)
1827 --------------------------------
1828 * The Status command now takes an optional list of variables to display
1829 along with channel status.
1830 * The QueueEntry event now also includes the channel's uniqueid
1834 * res_odbc no longer has a limit of 1023 total possible unshared connections,
1835 as some people were running into this limit. This limit has been increased
1840 * The TRANSFER queue log entry now includes the the caller's original
1841 position in the transferred-from queue.
1842 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
1843 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
1844 as well as an explanation about timeout options in general
1845 * Added a new option - C - for forcing the "answered elsewhere" flag on
1846 cancellation of calls in to members of the queue. This is to avoid the
1847 call to a member of a queue having the call listed as a "missed call".
1851 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
1852 adaptive capabilities. What this means in practical terms is that if your
1853 realtime table lacks critical fields, Asterisk will now emit warnings to
1854 that effect. Also, some of the realtime drivers have the ability (if
1855 configured) to automatically add those columns to the table with the
1856 correct type and length.
1860 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
1861 the 'setvar' option to cause a given audio file to be played upon completion
1862 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
1863 Skinny channels only.
1864 * You can now compile Asterisk against the Hoard Memory Allocator, see the
1865 Hoard page on the Asterisk wiki for more information:
1866 https://wiki.asterisk.org/wiki/x/pQBB
1867 * Config file variables may now be appended to, by using the '+=' append
1868 operator. This is most helpful when working with long SQL queries in
1869 func_odbc.conf, as the queries no longer need to be specified on a single
1871 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
1872 which will add a second to the billsec when the ending
1873 time is set, if the number in the microseconds field of the end time is
1874 greater than the number of microseconds in the answer time. This allows
1875 users to count the 'initiated' seconds in their billing records.
1877 ------------------------------------------------------------------------------
1878 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
1879 ------------------------------------------------------------------------------
1881 AMI - The manager (TCP/TLS/HTTP)
1882 --------------------------------
1883 * Manager has undergone a lot of changes, all of them documented
1884 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
1885 * Manager version has changed to 1.1
1886 * Added a new action 'CoreShowChannels' to list currently defined channels
1887 and some information about them.
1888 * Added a new action 'SIPshowregistry' to list SIP registrations.
1889 * Added TLS support for the manager interface and HTTP server
1890 * Added the URI redirect option for the built-in HTTP server
1891 * The output of CallerID in Manager events is now more consistent.
1892 CallerIDNum is used for number and CallerIDName for name.
1893 * Enable https support for builtin web server.
1894 See configs/http.conf.sample for details.
1895 * Added a new action, GetConfigJSON, which can return the contents of an
1896 Asterisk configuration file in JSON format. This is intended to help
1897 improve the performance of AJAX applications using the manager interface
1899 * SIP and IAX manager events now use "ChannelType" in all cases where we
1900 indicate channel driver. Previously, we used a mixture of "Channel"
1901 and "ChannelDriver" headers.
1902 * Added a "Bridge" action which allows you to bridge any two channels that
1903 are currently active on the system.
1904 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
1905 the voicemail users setup.
1906 * Added 'DBDel' and 'DBDelTree' manager commands.
1907 * cdr_manager now reports events via the "cdr" level, separating it from
1908 the very verbose "call" level.
1909 * Manager users are now stored in memory. If you change the manager account
1910 list (delete or add accounts) you need to reload manager.
1911 * Added Masquerade manager event for when a masquerade happens between
1913 * Added "manager reload" command for the CLI
1914 * Lots of commands that only provided information are now allowed under the
1915 Reporting privilege, instead of only under Call or System.
1916 * The IAX* commands now require either System or Reporting privilege, to
1917 mirror the privileges of the SIP* commands.
1918 * Added ability to retrieve list of categories in a config file.
1919 * Added ability to retrieve the content of a particular category.
1920 * Added ability to empty a context.
1921 * Created new action to create a new file.
1922 * Updated delete action to allow deletion by line number with respect to category.
1923 * Added new action insert to add new variable to category at specified line.
1924 * Updated action newcat to allow new category to be inserted in file above another
1926 * Added new event "JitterBufStats" in the IAX2 channel
1927 * Originate now requires the Originate privilege and, if you want to call out
1928 to a subshell, it requires the System privilege, as well. This was done to
1929 enhance manager security.
1930 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
1931 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
1932 or manager show command Atxfer from the CLI
1933 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
1934 details or manager show command IAXregistry from the CLI
1938 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
1939 state in the dialplan, as well as creating custom device states that are
1940 controllable from the dialplan.
1941 * Extend CALLERID() function with "pres" and "ton" parameters to
1942 fetch string representation of calling number presentation indicator
1943 and numeric representation of type of calling number value.
1944 * MailboxExists converted to dialplan function
1945 * A new option to Dial() for telling IP phones not to count the call
1946 as "missed" when dial times out and cancels.
1947 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
1948 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
1949 held for any given channel. Also, locks are automatically freed when a
1951 * Added HINT() dialplan function that allows retrieving hint information.
1952 Hints are mappings between extensions and devices for the sake of
1953 determining the state of an extension. This function can retrieve the list
1954 of devices or the name associated with a hint.
1955 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
1957 * Added SYSINFO() dialplan function which allows retrieval of system information
1958 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
1959 the existence of a dialplan target.
1960 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
1961 upper and lower case, respectively.
1962 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
1963 ID for the call (not the Asterisk call ID or unique ID), provided that the
1964 channel driver supports this. For SIP, you get the SIP call-ID for the
1965 bridged channel which you can store in the CDR with a custom field.
1969 * Added CLI permissions, config file: cli_permissions.conf
1970 default is to allow all commands for every local user/group.
1971 Also this new feature added three new CLI commands:
1972 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
1973 - cli reload permissions
1974 - cli show permissions
1975 * New CLI command "core show hint" (usage: core show hint <exten>)
1976 * New CLI command "core show settings"
1977 * Added 'core show channels count' CLI command.
1978 * Added the ability to set the core debug and verbose values on a per-file basis.
1979 * Added 'queue pause member' and 'queue unpause member' CLI commands
1980 * Ability to set process limits ("ulimit") without restarting Asterisk
1981 * Enhanced "agi debug" to print the channel name as a prefix to the debug
1982 output to make debugging on busy systems much easier.
1983 * New CLI commands "dialplan set extenpatternmatching true/false"
1984 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
1985 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
1986 listed in the startup_commands section of cli.conf will get executed.
1987 * Added a CLI command, "devstate change", which allows you to set custom device
1988 states from the func_devstate module that provides the DEVICE_STATE() function
1989 and handling of the "Custom:" devices.
1990 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
1991 sorted into the different possible callbacks, with the number of entries
1992 currently scheduled for each. Gives you a feel for how busy the sip channel
1994 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
1995 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
1996 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
2000 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
2001 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
2002 for a received call. If it is detected, the channel will jump to the
2003 'fax' extension in the dialplan.
2004 * The default SIP useragent= identifier now includes the Asterisk version
2005 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
2006 If set, and the incoming request carries authentication info,
2007 the username to match in the users list is taken from the Digest header
2008 rather than from the From: field. This feature is considered experimental.
2009 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
2010 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
2011 * The "localmask" setting was removed in version 1.2 and the reminder about it
2012 being removed is now also removed.
2013 * A new option "busylevel" for setting a level of calls where asterisk reports
2014 a device as busy, to separate it from call-limit. This value is also added
2015 to the SIP_PEER dialplan function.
2016 * A new realtime family called "sipregs" is now supported to store SIP registration
2017 data. If this family is defined, "sippeers" will be used for configuration and
2018 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
2019 registration data, as before.
2020 * The SIPPEER function have new options for port address, call and pickup groups
2021 * Added support for T.140 realtime text in SIP/RTP
2022 * The "checkmwi" option has been removed from sip.conf, as it is no longer
2023 required due to the restructuring of how MWI is handled. See the descriptions
2024 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
2025 for more information.
2026 * Added rtpdest option to CHANNEL() dialplan function.
2027 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
2028 * SIP now adds a header to the CANCEL if the call was answered by another phone
2029 in the same dial command, or if the new c option in dial() is used.
2030 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
2031 states it is not needed. For phones, however, that do require it the "registertrying" option
2032 has been added so it can be enabled.
2033 * A new option called "callcounter" (global/peer/user level) enables call counters needed
2034 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
2035 used to enable this functionality).
2036 * New settings for timer T1 and timer B on a global level or per device. This makes it
2037 possible to force timeout faster on non-responsive SIP servers. These settings are
2038 considered advanced, so don't use them unless you have a problem.
2039 * Added a dial string option to be able to set the To: header in an INVITE to any
2041 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
2042 the qualify frequency.
2043 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
2044 were not properly torn down due to network or endpoint failures during an established
2046 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
2047 and configs/sip.conf.sample for more information on how it is used.
2048 * Added a new configuration option "authfailureevents" that enables manager events when
2049 a peer can't authenticate properly.
2050 * Added DNS manager support to registrations for peers not referencing a peer entry.
2054 * Added the trunkmaxsize configuration option to chan_iax2.
2055 * Added the srvlookup option to iax.conf
2056 * Added support for OSP. The token is set and retrieved through the CHANNEL()
2059 XMPP Google Talk/Jingle changes
2060 -------------------------------
2061 * Added the bindaddr option to gtalk.conf.
2065 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
2066 * Proper codec support in chan_skinny.
2067 * Added settings for IP and Ethernet QoS requests
2071 * Added separate settings for media QoS in mgcp.conf
2073 Console Channel Driver changes
2074 ------------------------------
2075 * Added experimental support for video send & receive to chan_oss.
2076 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
2079 Phone channel changes (chan_phone)
2080 ----------------------------------
2081 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
2083 H.323 channel Changes
2084 ---------------------
2085 * H323 remote hold notification support added (by NOTIFY message
2086 and/or H.450 supplementary service)
2088 Local channel changes
2089 ---------------------
2090 * The device state functionality in the Local channel driver has been updated
2091 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
2092 to just UNKNOWN if the extension exists.
2093 * Added jitterbuffer support for chan_local. This allows you to use the
2094 generic jitterbuffer on incoming calls going to Asterisk applications.
2095 For example, this would allow you to use a jitterbuffer for an incoming
2096 SIP call to Voicemail by putting a Local channel in the middle. This
2097 feature is enabled by using the 'j' option in the Dial string to the Local
2098 channel in conjunction with the existing 'n' option for local channels.
2099 * A 'b' option has been added which causes chan_local to return the actual channel
2100 that is behind it when queried. This is useful for transfer scenarios as the
2101 actual channel will be transferred, not the Local channel.
2103 Agent channel changes
2104 ----------------------
2105 * The ackcall and endcall options are now supplemented with options acceptdtmf
2106 and enddtmf. These allow for the DTMF keypress to be configurable. The options
2107 default to their old hard-coded values ('#' and '*' respectively) so this should
2108 not break any existing agent installations.
2110 DAHDI channel driver (chan_dahdi) Changes
2111 ----------------------------------------
2112 * SS7 support (via libss7 library)
2113 * In India, some carriers transmit CID via dtmf. Some code has been added
2114 that will handle some situations. The cidstart=polarity_IN choice has been added for
2115 those carriers that transmit CID via dtmf after a polarity change.
2116 * CID matching information is now shown when doing 'dialplan show'.
2117 * Added dahdi show version CLI command.
2118 * Added setvar support to chan_dahdi.conf channel entries.
2119 * Added two new options: mwimonitor and mwimonitornotify. These options allow
2120 you to enable MWI monitoring on FXO lines. When the MWI state changes,
2121 the script specified in the mwimonitornotify option is executed. An internal
2122 event indicating the new state of the mailbox is also generated, so that
2123 the normal MWI facilities in Asterisk work as usual.
2124 * Added signalling type 'auto', which attempts to use the same signalling type
2125 for a channel as configured in DAHDI. This is primarily designed for analog
2126 ports, but will also work for digital ports that are configured for FXS or FXO
2127 signalling types. This mode is also the default now, so if your chan_dahdi.conf
2128 does not specify signalling for a channel (which is unlikely as the sample
2129 configuration file has always recommended specifying it for every channel) then
2130 the 'auto' mode will be used for that channel if possible.
2131 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
2132 state for a channel; also ensured that the DNDState Manager event is
2133 emitted no matter how the DND state is set or cleared.
2137 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
2138 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
2139 for details. This new channel driver allows you to use Nortel i2002,
2140 i2004, and i2050 phones with Asterisk.
2141 * Added a new channel driver, chan_console, which uses portaudio as a cross
2142 platform audio interface. It was written as a channel driver that would
2143 work with Mac CoreAudio, but portaudio supports a number of other audio
2144 interfaces, as well. Note that this channel driver requires v19 or higher
2145 of portaudio; older versions have a different API.
2149 * Added the ability to specify arguments to the Dial application when using
2150 the DUNDi switch in the dialplan.
2151 * Added the ability to set weights for responses dynamically. This can be
2152 done using a global variable or a dialplan function. Using the SHELL()
2153 function would allow you to have an external script set the weight for
2155 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
2156 functions will allow you to initiate a DUNDi query from the dialplan,
2157 find out how many results there are, and access each one.
2158 * Added the ability to specifiy a port for a dundi peer.
2162 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
2163 functions will allow you to initiate an ENUM lookup from the dialplan,
2164 and Asterisk will cache the results. ENUMRESULT can be used to access
2165 the results without doing multiple DNS queries.
2169 * Added the ability to customize which sound files are used for some of the
2170 prompts within the Voicemail application by changing them in voicemail.conf
2171 * Added the ability for the "voicemail show users" CLI command to show users
2172 configured by the dynamic realtime configuration method.
2173 * MWI (Message Waiting Indication) handling has been significantly
2174 restructured internally to Asterisk. It is now totally event based
2175 instead of polling based. The voicemail application will notify other
2176 modules that have subscribed to MWI events when something in the mailbox
2178 This also means that if any other entity outside of Asterisk is changing
2179 the contents of mailboxes, then the voicemail application still needs to
2180 poll for changes. Examples of situations that would require this option
2181 are web interfaces to voicemail or an email client in the case of using
2182 IMAP storage. So, two new options have been added to voicemail.conf
2183 to account for this: "pollmailboxes" and "pollfreq". See the sample
2184 configuration file for details.
2185 * Added "tw" language support
2186 * Added support for storage of greetings using an IMAP server
2187 * Added ability to customize forward, reverse, stop, and pause keys for message playback
2188 * SMDI is now enabled in voicemail using the smdienable option.
2189 * A "lockmode" option has been added to asterisk.conf to configure the file
2190 locking method used for voicemail, and potentially other things in the
2191 future. The default is the old behavior, lockfile. However, there is a
2192 new method, "flock", that uses a different method for situations where the
2193 lockfile will not work, such as on SMB/CIFS mounts.
2194 * Added the ability to backup deleted messages, to ease recovery in the case
2195 that a user accidentally deletes a message, and discovers that they need it.
2196 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
2197 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
2198 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
2199 voicemail boxes. The SMDI interface can also poll for MWI changes when some
2200 outside entity is modifying the state of the mailbox (such as IMAP storage or
2201 a web interface of some kind).
2202 * Added the support for marking messages as "urgent." There are two methods to accomplish
2203 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
2204 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
2205 the message as urgent after he has recorded a voicemail by following the voice instructions.
2206 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
2211 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
2212 used across multiple queues.
2213 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
2214 setqueueentryvar options for each queue, see queues.conf.sample for details.
2215 * Added keepstats option to queues.conf which will keep queue
2216 statistics during a reload.
2217 * setinterfacevar option in queues.conf also now sets a variable
2218 called MEMBERNAME which contains the member's name.
2219 * Added 'Strategy' field to manager event QueueParams which represents
2220 the queue strategy in use.
2221 * Added option to run macro when a queue member is connected to a caller,
2222 see queues.conf.sample for details.
2223 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
2224 does not count paused queue members as unavailable.
2225 * Added min-announce-frequency option to queues.conf which allows you to control the
2226 minimum amount of time between queue announcements for use when the caller's queue
2227 position changes frequently.
2228 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
2230 * Added ability for non-realtime queues to have realtime members
2231 * Added the "linear" strategy to queues.
2232 * Added the "wrandom" strategy to queues.
2233 * Added new channel variable QUEUE_MIN_PENALTY
2234 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
2235 rules in queuerules.conf. See configs/queuerules.conf.sample for details
2236 * Added a new parameter for member definition, called state_interface. This may be
2237 used so that a member may be called via one interface but have a different interface's
2238 device state reported.
2239 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
2240 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
2241 "manager show command QueueReset."
2242 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
2243 specified by the periodic-announce option, then one will be chosen randomly when it is time
2244 to play a periodic announcment
2245 * New configuration options: announce-position now takes two more values in addition to "yes" and
2246 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
2247 announce-position-limit. By setting announce-position to "limit" callers will only have their
2248 position announced if their position is less than what is specified by announce-position-limit.
2249 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
2250 will be told that their are more than announce-position-limit callers waiting.
2251 * Two new queue log events have been added. An ADDMEMBER event will be logged
2252 when a realtime queue member is added and a REMOVEMEMBER event will be logged
2253 when a realtime queue member is removed. Since there is no calling channel associated
2254 with these events, the string "REALTIME" is placed where the channel's unique id
2255 is typically placed.
2256 * The configuration method for the "joinempty" and "leavewhenempty" options has
2257 changed to a comma-separated list of methods of determining member availability
2258 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
2259 values are still accepted for backwards-compatibility, though.
2260 * The average talktime is now calculated on queues. This information is reported via the
2261 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
2262 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
2267 * The 'o' option to provide an optimization has been removed and its functionality
2268 has been enabled by default.
2269 * When a conference is created, the UNIQUEID of the channel that caused it to be
2270 created is stored. Then, every channel that joins the conference will have the
2271 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
2272 callers that come and go from long standing conferences.
2273 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
2274 except it does operations on a channel by name, instead of number in a conference.
2275 This is a very useful feature in combination with the 'X' option to ChanSpy.
2276 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
2278 * Added new RealTime functionality to provide support for scheduled conferencing.
2279 This includes optional messages to the caller if they attempt to join before
2280 the schedule start time, or to allow the caller to join the conference early.
2281 Also included is optional support for limiting the number of callers per
2282 RealTime conference.
2283 * Added the S() and L() options to the MeetMe application. These are pretty
2284 much identical to the S() and L() options to Dial(). They let you set
2285 timeouts for the conference, as well as have warning sounds played to
2286 let the caller know how much time is left, and when it is running out.
2287 * Added the ability to do "meetme concise" with the "meetme" CLI command.
2288 This extends the concise capabilities of this CLI command to include
2289 listing all conferences, instead of an addition to the other sub commands
2290 for the "meetme" command.
2291 * Added the ability to specify the music on hold class used to play into the
2292 conference when there is only one member and the M option is used.
2293 * Added MEETME_INFO dialplan function which provides a way to query
2294 various properties of a Meetme conference.
2295 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
2296 and *84: record in-conf
2298 Other Dialplan Application Changes
2299 ----------------------------------
2300 * Argument support for Gosub application
2301 * From the to-do lists: straighten out the app timeout args:
2302 Wait() app now really does 0.3 seconds- was truncating arg to an int.
2303 WaitExten() same as Wait().
2304 Congestion() - Now takes floating pt. argument.
2305 Busy() - now takes floating pt. argument.
2306 Read() - timeout now can be floating pt.
2307 WaitForRing() now takes floating pt timeout arg.
2308 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
2309 * Added 's' option to Page application.
2310 * Added an optional timeout argument to the Page application.
2311 * Added 'E', 'V', and 'P' commands to ExternalIVR.
2312 * Added 'o' and 'X' options to Chanspy.
2313 * Added a new dialplan application, Bridge, which allows you to bridge the
2314 calling channel to any other active channel on the system.
2315 * Added the ability to specify a music on hold class to play instead of ringing
2316 for the SLATrunk application.
2317 * The Read application no longer exits the dialplan on error. Instead, it sets
2318 READSTATUS to ERROR, which you can catch and handle separately.
2319 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
2320 of asking for verification of each name, one at a time.
2321 * Privacy() no longer uses privacy.conf, as all options are specifyable as
2322 direct options to the app.
2323 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
2325 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
2326 * The ChannelRedirect application no longer exits the dialplan if the given channel
2327 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
2328 or NOCHANNEL if the given channel was not found.
2329 * The silencethreshold setting that was previously configurable in multiple
2330 applications is now settable globally via dsp.conf.
2332 Music On Hold Changes
2333 ---------------------
2334 * A new option, "digit", has been added for music on hold classes in
2335 musiconhold.conf. If this is set for a music on hold class, a caller
2336 listening to music on hold can press this digit to switch to listening
2337 to this music on hold class.
2338 * Support for realtime music on hold has been added.
2339 * In conjunction with the realtime music on hold, a general section has
2340 been added to musiconhold.conf, its sole variable is cachertclasses. If this
2341 is set, then music on hold classes found in realtime will be cached in memory.
2345 * AEL upgraded to use the Gosub with Arguments instead
2346 of Macro application, to hopefully reduce the problems
2347 seen with the artificially low stack ceiling that
2348 Macro bumps into. Macros can only call other Macros
2349 to a depth of 7. Tests run using gosub, show depths
2350 limited only by virtual memory. A small test demonstrated
2351 recursive call depths of 100,000 without problems.
2352 -- in addition to this, all apps that allowed a macro
2353 to be called, as in Dial, queues, etc, are now allowing
2354 a gosub call in similar fashion.
2355 * AEL now generates LOCAL(argname) declarations when it
2356 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
2357 etc. That makes the arguments local in scope. The user
2358 can define their own local variables in macros, now,
2359 by saying "local myvar=someval;" or using Set() in this
2360 fashion: Set(LOCAL(myvar)=someval); ("local" is now
2362 * utils/conf2ael introduced. Will convert an extensions.conf
2363 file into extensions.ael. Very crude and unfinished, but
2364 will be improved as time goes by. Should be useful for a
2365 first pass at conversion.
2366 * aelparse will now read extensions.conf to see if a referenced
2367 macro or context is there before issueing a warning.
2368 * AEL parser sets a local channel variable ~~EXTEN~~, to
2369 preserve the value of ${EXTEN} thru switch statements.
2370 * New operator in $[...] expressions: the ~~ operator serves
2371 as a concatenation operator. AT THE MOMENT, it is really only
2372 necessary and useful in AEL, especially in if() expressions.
2373 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
2374 any enclosing double-quotes, and evaluate to the value of a
2375 concatenated with the value of b. For example if a is set to
2376 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
2377 evaluate to xyzabc .
2380 Call Features (res_features) Changes
2381 ------------------------------------
2382 * Added the parkedcalltransfers option to features.conf
2383 * Added parkedcallparking option to control one touch parking w/ parking
2385 * Added parkedcallhangup option to control disconnect feature w/ parking
2387 * Added parkedcallrecording option to control one-touch record w/ parking
2389 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
2390 parkedcalltransfers option support for multiple parking lots.
2391 * Added BRIDGE_FEATURES variable to set available features for a channel
2392 * The built-in method for doing attended transfers has been updated to
2393 include some new options that allow you to have the transferee sent
2394 back to the person that did the transfer if the transfer is not successful.
2395 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
2396 in features.conf.sample.
2397 * Added support for configuring named groups of custom call features in
2398 features.conf. This means that features can be written a single time, and
2399 then mapped into groups of features for different key mappings or easier
2401 * Updated the ParkedCall application to allow you to not specify a parking
2402 extension. If you don't specify a parking space to pick up, it will grab
2403 the first one available.
2404 * Added cli command 'features reload' to reload call features from features.conf
2405 * Moved into core asterisk binary.
2406 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
2407 * Added the ability for custom parking lots to be configured with their own
2408 parking extension with the parkext option.
2410 Language Support Changes
2411 ------------------------
2412 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
2413 * Added support for the Hungarian language for saying numbers, dates, and times.
2417 * Added SPEECH commands for speech recognition. A complete listing can be found
2419 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
2420 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
2421 does not behave as expected; the native command needs to be used, instead.
2422 * Added the ability to perform SRV lookups on fast AGI calls. To use this
2423 feature, simply use hagi: instead of agi: as the protocol portion
2424 of the URI parameter to the AGI function call in your dial plan. Also note
2425 that specifying a port number in the AGI URI will disable SRV lookups,
2426 even if you use the hagi: protocol.
2427 * No longer support MSG_OOB flag on HANGUP.
2431 * Added rotatestrategy option to logger.conf, along with two new options:
2432 "timestamp" which will use the time to name the logger files instead of
2433 sequence number; and "rotate", which rotates the names of the log files,
2434 similar to the way syslog rotates files.
2435 * Added exec_after_rotate option to logger.conf, which allows a system
2436 command to be run after rotation. This is primarily useful with
2437 rotatestrategy=rotate, to allow a limit on the number of log files kept
2438 and to ensure that the oldest log file gets deleted.
2439 * Added realtime support for the queue log
2443 * The cdr_manager module has a [mappings] feature, like cdr_custom,
2444 to add fields to the manager event from the CDR variables.
2445 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
2446 backend database CDR table. Specifically, additional, non-standard
2447 columns are supported, merely by setting the corresponding CDR variable in
2448 your dialplan. In addition, you may alias any column to another name (for
2449 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
2450 simply "alias src => ANI" in the configuration file). Records may be
2451 posted to more than one backend, simply by specifying multiple categories
2452 in the configuration file. And finally, you may filter which CDRs get
2453 posted to each backend, by specifying a filter (which the record must
2454 match) for the particular category. Filters are additive (meaning all
2455 rules must match to post that CDR).
2456 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
2457 module. Specifically, you may add additional columns into the table and
2458 they will be set, if you set the corresponding CDR variable name. Also,
2459 if you omit columns in your database table, they will be silently skipped
2460 (but a record will still be inserted, based on what columns remain). Note
2461 that the other two features from cdr_adaptive_odbc (alias and filter) are
2462 not currently supported.
2463 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
2464 has been disabled using the NoCDR application.
2466 Miscellaneous New Modules
2467 -------------------------
2468 * Added a new CDR module, cdr_sqlite3_custom.
2469 * Added a new realtime configuration module, res_config_sqlite
2470 * Added a new codec translation module, codec_resample, which re-samples
2471 signed linear audio between 8 kHz and 16 kHz to help support wideband
2473 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
2474 based on configuration templates that use Asterisk dialplan function and
2475 variable substitution. It should be possible to create phone profiles and
2476 templates that work for the majority of phones provisioned over http. It
2477 is currently only intended to provision a single user account per phone.
2478 An example profile and set of templates for Polycom phones is provided.
2479 NOTE: Polycom firmware is not included, but should be placed in
2480 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
2481 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
2482 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
2483 provided; there is a JACK() application, and a JACK_HOOK() function. Both
2484 interfaces create an input and output JACK port. The application makes
2485 these ports the endpoint of the call. The audio coming from the channel
2486 goes out the output port and whatever comes back in on the input port is
2487 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
2488 audiohook on the channel. This lets you run the audio coming from a
2489 channel through JACK, and whatever comes back in is what gets forwarded
2490 on as the channel's audio. This is very useful for building custom
2491 vocoders or doing recording or analysis of the channel's audio in another
2493 * Added a new module, res_config_curl, which permits using a HTTP POST url
2494 to retrieve, create, update, and delete realtime information from a remote
2495 web server. Note that this module requires func_curl.so to be loaded for
2496 backend functionality.
2497 * Added a new module, res_config_ldap, which permits the use of an LDAP
2498 server for realtime data access.
2499 * Added support for writing and running your dialplan in lua using the pbx_lua
2500 module. See configs/extensions.lua.sample for examples of how to do this.
2504 * Ability to use libcap to set high ToS bits when non-root
2505 on Linux. If configure is unable to find libcap then you
2506 can use --with-cap to specify the path.
2507 * Added maxfiles option to options section of asterisk.conf which allows you to specify
2508 what Asterisk should set as the maximum number of open files when it loads.
2509 * Added the jittertargetextra configuration option.
2510 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
2511 configuration files for the IP channel drivers. The new option is "cos".
2512 This information is also documented on the Asterisk wiki at
2513 https://wiki.asterisk.org/wiki/x/EYBG
2514 * When originating a call using AMI or pbx_spool that fails the reason for failure
2515 will now be available in the failed extension using the REASON dialplan variable.
2516 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
2517 It allows you to configure a prefix for auto-monitor recordings.
2518 * A new extension pattern matching algorithm, based on a trie, is introduced
2519 here, that could noticeably speed up mid-sized to large dialplans.
2520 It is NOT used by default, as duplicating the behaviour of the old pattern
2521 matcher is still under development. A config file option, in extensions.conf,
2522 in the [general] section, called "extenpatternmatchingnew", is by default
2523 set to false; setting that to true will force the use of the new algorithm.
2524 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
2525 be used to switch the algorithms at run time.
2526 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
2527 specifying which socket to use to connect to the running Asterisk daemon
2529 * Performance enhancements to the sched facility, which is used in
2530 the channel drivers, etc. Added hashtabs and doubly-linked lists
2531 to speed up deletion; start at the beginning or end of list to
2533 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
2534 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
2535 Added regression tests to the tests/ dir, also.
2536 * Added a refcount trace feature to astobj2 for those trying to balance
2537 object creation, deletion; work, play; space and time. See the
2538 notes in astobj2.h. Also, see utils/refcounter as well, as a
2539 quick way to find unbalanced refcounts in what could be a sea
2540 of objects that were balanced.
2541 * Added logging to 'make update' command. See update.log
2542 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
2543 do not come from the remote party.
2544 * Added the 'n' option to the SpeechBackground application to tell it to not
2545 answer the channel if it has not already been answered.
2546 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
2547 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
2549 * iLBC source code no longer included (see UPGRADE.txt for details)
2550 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
2551 deadlock is detected, a backtrace of the stack which led to the lock calls
2552 will be output to the CLI.
2553 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
2554 the "core show locks" CLI command will give lock information output as well
2555 as a backtrace of the stack which led to the lock calls.
2556 * users.conf now sports an optional alternateexts property, which permits
2557 allocation of additional extensions which will reach the specified user.
2558 * A new option for the configure script, --enable-internal-poll, has been added
2559 for use with systems which may have a buggy implementation of the poll system
2560 call. If you notice odd behavior such as the CLI being unresponsive on remote
2561 consoles, you may want to try using this option. This option is enabled by default
2562 on Darwin systems since it is known that the Darwin poll() implementation has
2566 --------------------
2567 * In addition to timing from DAHDI, there is a new timing module called
2568 res_timing_timerfd. In order to use this, you must be running Linux with
2569 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
2570 script will be able to tell if you have the requirements. From menuselect, select
2571 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob