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 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
383 items (separated by commas), and items in the rule can be negated by prefixing
384 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
385 longer necessray to control the order that the 'permit' and 'deny' columns are
386 returned from queries.
388 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
389 be used within the dynamic weight attribute when specifying a mapping.
391 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
392 header, instead of putting the user defined event name there. When enabled
393 the UserDefType header is added for user defined events. This feature is
394 enabled with the setting show_user_defined.
396 * Macro has been deprecated in favor of GoSub. For redirecting and connected
397 line purposes use the following variables instead of their macro equivalents:
398 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
399 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
400 cc_callback_macro in channel configurations.
405 * A new channel variable, AGIEXITONHANGUP, has been added which allows
406 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
407 AGI application would exit immediately after a channel hangup is detected.
409 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
410 are resolved and each address is attempted in turn until one succeeds or
414 AMI (Asterisk Manager Interface)
416 * Originate now generates an error response if the extension given is not found
419 * MixMonitor will now show IDs associated with the mixmonitor upon creating
420 them if the i(variable) option is used. StopMixMonitor will accept
421 MixMonitorID as an option to close specific MixMonitors.
423 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
424 updated to include information about peers configured with
425 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
426 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
427 returned if auto_force_rport is not enabled.
429 * Hangup now can take a regular expression as the Channel option. If you want
430 to hangup multiple channels, use /regex/ as the Channel option. Existing
431 behavior to hanging up a single channel is unchanged, but if you pass a regex,
432 the manager will send you a list of channels back that were hung up.
434 * Support for IPv6 addresses has been added.
436 * AMI Events can now be documented in the Asterisk source. Note that AMI event
437 documentation is only generated when Asterisk is compiled using 'make full'.
438 See the CLI section for commands to display AMI event information.
440 * The AMI Hangup event now includes the AccountCode header so you can easily
441 correlate with AMI Newchannel events.
443 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
444 the StateInterface of the queue member.
446 * Added AMI event SessionTimeout in the Call category that is issued when a
447 call is terminated due to either RTP stream inactivity or SIP session timer
450 * CEL events can now contain a user defined header UserDefType. See core
451 changes for more information.
453 * OOH323 ChannelUpdate events now contain a CallRef header.
455 * Added PresenceState command. This command will report the presence state for
456 the given presence provider.
458 * Added Parkinglots command. This will list all parking lots as a series of
459 AMI Parkinglot events.
461 * Added MessageSend command. This behaves in the same manner as the
462 MessageSend application, and is a technolgoy agnostic mechanism to send out
463 of call text messages.
465 * Added "message" class authorization. This grants an account permission to
466 send out of call messages. Write-only.
471 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
472 filenames of all running mixmonitors on a channel.
474 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
475 numeric instead of 0, 1, or 2.
477 * "stun show status" will show a table describing how the STUN client is
480 * "acl show [named acl]" will show information regarding a Named ACL. The
481 acl module can be reloaded with "reload acl".
483 * Added CLI command to display AMI event information - "manager show events",
484 which shows a list of all known and documented AMI events, and "manager show
485 event [event name]", which shows detail information about a specific AMI
488 * The result of the CLI command "queue show" now includes the state interface
489 information of the queue member.
491 * The command "core set verbose" will now set a separate level of logging for
492 each remote console without affecting any other console.
494 * Added command "cdr show pgsql status" to check connection status
496 * "sip show channel" will now display the complete route set.
498 * Added "presencestate list" command. This command will list all custom
499 presence states that have been set by using the PRESENCE_STATE dialplan
502 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
503 command. This changes a custom presence to a new state.
508 * Codec lists may now be modified by the '!' character, to allow succinct
509 specification of a list of codecs allowed and disallowed, without the
510 requirement to use two different keywords. For example, to specify all
511 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
513 * Add support for parsing SDP attributes, generating SDP attributes, and
514 passing it through. This support includes codecs such as H.263, H.264, SILK,
515 and CELT. You are able to set up a call and have attribute information pass.
516 This should help considerably with video calls.
518 * The iLBC codec can now use a system-provided iLBC library if one is installed,
519 just like the GSM codec.
523 * Asterisk version and build information is now logged at the beginning of a
526 * Threads belonging to a particular call are now linked with callids which get
527 added to any log messages produced by those threads. Log messages can now be
528 easily identified as involved with a certain call by looking at their call id.
529 Call ids may also be attached to log messages for just about any case where
530 it can be determined to be related to a particular call.
532 * Each logging destination and console now have an independent notion of the
533 current verbosity level. Logger.conf now allows an optional argument to
534 the 'verbose' specifier, indicating the level of verbosity sent to that
535 particular logging destination. Additionally, remote consoles now each
536 have their own verbosity level. The command 'core set verbose' will now set
537 a separate level for each remote console without affecting any other
543 * Added 'announcement' option which will play at the start of MOH and between
544 songs in modes of MOH that can detect transitions between songs (eg.
550 * New per parking lot options: comebackcontext and comebackdialtime. See
551 configs/features.conf.sample for more details.
553 * Channel variable PARKER is now set when comebacktoorigin is disabled in
556 * Channel variable PARKEDCALL is now set with the name of the parking lot
557 when a timeout occurs.
563 CDR Postgresql Driver
565 * Added command "cdr show pgsql status" to check connection status
568 CDR Adaptive ODBC Driver
570 * Added schema option for databases that support specifying a schema.
578 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
579 CALENDAR_WRITE has completed successfully.
584 * A new option, 'probation' has been added to rtp.conf
585 RTP in strictrtp mode can now require more than 1 packet to exit learning
586 mode with a new source (and by default requires 4). The probation option
587 allows the user to change the required number of packets in sequence to any
588 desired value. Use a value of 1 to essentially restore the old behavior.
589 Also, with strictrtp on, Asterisk will now drop all packets until learning
590 mode has successfully exited. These changes are based on how pjmedia handles
591 media sources and source changes.
593 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
594 enabled or disabled using the icesupport setting. A variety of other
595 settings have been introduced to configure STUN/TURN connections.
600 * A new module, res_corosync, has been introduced. This module uses the
601 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
602 of Asterisk servers to both Message Waiting Indication (MWI) and/or
603 Device State (presence) information. This module is very similar to, and
604 is a replacement for the res_ais module that was in previous releases of
610 * This module adds a cleaned up, drop-in replacement for res_jabber called
611 res_xmpp. This provides the same externally facing functionality but is
612 implemented differently internally. res_jabber has been deprecated in favor
613 of res_xmpp; please see the UPGRADE.txt file for more information.
618 * The safe_asterisk script has been updated to allow several of its parameters
619 to be set from environment variables. This also enables a custom run
620 directory of Asterisk to be specified, instead of defaulting to /tmp.
622 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
623 its value to determine the directory to assume is the top-level directory of
624 the source tree. If the variable is not set, it defaults to the current
625 behavior and uses the current working directory.
628 ------------------------------------------------------------------------------
629 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
630 ------------------------------------------------------------------------------
634 * Asterisk now has protocol independent support for processing text messages
635 outside of a call. Messages are routed through the Asterisk dialplan.
636 SIP MESSAGE and XMPP are currently supported. There are options in
637 jabber.conf and sip.conf to allow enabling these features.
638 -> jabber.conf: see the "sendtodialplan" and "context" options.
639 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
640 and "outofcall_message_context" options.
641 The MESSAGE() dialplan function and MessageSend() application have been
642 added to go along with this functionality. More detailed usage information
643 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
644 * If real-time text support (T.140) is negotiated, it will be preferred for
645 sending text via the SendText application. For example, via SIP, messages
646 that were once sent via the SIP MESSAGE request would be sent via RTP if
647 T.140 text is negotiated for a call.
651 * parkedmusicclass can now be set for non-default parking lots.
653 Asterisk Manager Interface
654 --------------------------
655 * PeerStatus now includes Address and Port.
656 * Added Hold events for when the remote party puts the call on and off hold
657 for chan_dahdi ISDN channels.
658 * Added new action MeetmeListRooms to list active conferences (shows same
659 data as "meetme list" at the CLI).
660 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
661 Description field that is set by 'description' in the channel configuration
663 * Added Uniqueid header to UserEvent.
664 * Added new action FilterAdd to control event filters for the current session.
665 This requires the system permission and uses the same filter syntax as
666 filters that can be defined in manager.conf
667 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
668 versions had some instances of the event converted, but others were left
669 as-is. All Unlink events should now be converted to Bridge events. The AMI
670 protocol version number was incremented to 1.2 as a result of this change.
673 --------------------------
674 * The HTTP Server can bind to IPv6 addresses.
677 --------------------------
678 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
679 with busydetect. usage example: busypattern=200,200,200,600
682 --------------------------
683 * New 'gtalk show settings' command showing the current settings loaded from
685 * The 'logger reload' command now supports an optional argument, specifying an
686 alternate configuration file to use.
687 * 'dialplan add extension' command will now automatically create a context if
688 the specified context does not exist with a message indicated it did so.
689 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
690 Description field which can be populated with 'description' in the channel
691 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
694 --------------------------
695 * The filter option in cdr_adaptive_odbc now supports negating the argument,
696 thus allowing records which do NOT match the specified filter.
697 * Added ability to log CONGESTION calls to CDR
700 --------------------------
701 * Ability to define custom SILK formats in codecs.conf.
702 * Addition of speex32 audio format with translation.
703 * CELT codec pass-through support and ability to define
704 custom CELT formats in codecs.conf.
705 * Ability to read raw signed linear files with sample rates
706 ranging from 8khz - 192khz. The new file extensions introduced
707 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
708 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
709 Skinny, H.323, etc) can still only support the following codecs:
710 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
711 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
712 Video: h261, h263, h263p, h264, mpeg4
717 --------------------------
718 * New highly optimized and customizable ConfBridge application capable of
719 mixing audio at sample rates ranging from 8khz-96khz.
720 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
721 and bridge profiles on a channel.
722 * CONFBRIDGE_INFO dialplan function capable of retrieving information
723 about a conference such as locked status and number of parties, admins,
725 * Addition of video_mode option in confbridge.conf for adding video support
726 into a bridge profile.
727 * Addition of the follow_talker video_mode in confbridge.conf. This video
728 mode dynamically switches the video feed to always display the loudest talker
729 supplying video in the conference.
733 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
734 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
735 variables from asterisk.conf.
739 * Addition of the JITTERBUFFER dialplan function. This function allows
740 for jitterbuffering to occur on the read side of a channel. By using
741 this function conference applications such as ConfBridge and MeetMe can
742 have the rx streams jitterbuffered before conference mixing occurs.
743 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
745 * Added STRREPLACE function. This function let's the user search a variable
746 for a given string to replace with another string as many times as the
747 user specifies or just throughout the whole string.
748 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
749 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
750 * Added extensions to chan_ooh323 in function CHANNEL()
752 libpri channel driver (chan_dahdi) DAHDI changes
753 --------------------------
754 * Added moh_signaling option to specify what to do when the channel's bridged
755 peer puts the ISDN channel on hold.
756 * Added display_send and display_receive options to control how the display ie
757 is handled. To send display text from the dialplan use the SendText()
758 application when the option is enabled.
759 * Added mcid_send option to allow sending a MCID request on a span.
762 --------------------------
763 * Added setvar option to calendar.conf to allow setting channel variables on
764 notification channels.
765 * Added "calendar show types" CLI command to list registered calendar
769 --------------------------
770 * Added two new options, r and t with file name arguments to record
771 single direction (unmixed) audio recording separate from the bidirectional
772 (mixed) recording. The mixed file name argument is optional now as long
773 as at least one recording option is used.
776 --------------------------
777 * Added a new option, l, which will disable local call optimization for
778 channels involved with the FollowMe thread. Use this option to improve
779 compatability for a FollowMe call with certain dialplan apps, options, and
783 --------------------------
784 * Added option "k" that will automatically close the conference when there's
785 only one person left when a user exits the conference.
788 --------------------------
789 * cel_pgsql now supports the 'extra' column for data added using the
790 CELGenUserEvent() application.
793 --------------------------
794 * Support for defining hints has been added to pbx_lua. See the 'hints' table
795 in the sample extensions.lua file for syntax details.
796 * Applications that perform jumps in the dialplan such as Goto will now
797 execute properly. When pbx_lua detects that the context, extension, or
798 priority we are executing on has changed it will immediately return control
799 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
800 the priority after the currently executing priority.
801 * An autoservice is now started by default for pbx_lua channels. It can be
802 stopped and restarted using the autoservice_stop() and autoservice_start()
806 --------------------------
807 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
808 into a FAXStatus event with an 'Operation' header that will be either
809 'send', 'receive', and 'gateway'.
810 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
811 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
812 feature will handle converting a fax call between an audio T.30 fax terminal
813 and an IFP T.38 fax terminal.
817 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
818 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
819 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
823 * Added general option negative_penalty_invalid default off. when set
824 members are seen as invalid/logged out when there penalty is negative.
825 for realtime members when set remove from queue will set penalty to -1.
826 * Added queue option autopausedelay when autopause is enabled it will be
827 delayed for this number of seconds since last successful call if there
828 was no prior call the agent will be autopaused immediately.
829 * Added member option ignorebusy this when set and ringinuse is not
830 will allow per member control of multiple calls as ringinuse does for
832 * Added global option check_state_unknown to enforce checking of device state
833 when the device state is unknown app_queue will see unknown as available.
837 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
839 * Added 'k' option to MeetMe to automatically kill the conference when there's only
840 one participant left (much like a normal call bridge)
841 * Added extra argument to Originate to set timeout.
845 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
846 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
847 utility in the UTILS section of menuselect. If an existing astdb is found and no
848 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
849 convert an existing astdb to the SQLite3 version automatically at runtime.
853 * Modules marked as deprecated are no longer marked as building by default. Enabling
854 these modules is still available via menuselect.
858 * authdebug is now disabled by default. To enable this functionaility again
859 set authdebug = yes in iax.conf.
863 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
864 releases it was disabled.
868 * The PBX core previously made a call with a non-existing extension test for
869 extension s@default and jump there if the extension existed.
870 This was a bad default behaviour and violated the principle of least surprise.
871 It has therefore been changed in this release. It may affect some
872 applications and configurations that rely on this behaviour. Most channel
873 drivers have avoided this for many releases by testing whether the extension
874 called exists before starting the PBX and generating a local error.
875 This behaviour still exists and works as before.
877 Extension "s" is used when no extension is given in a channel driver,
878 like immediate answer in DAHDI or calling to a domain with no user part
881 ------------------------------------------------------------------------------
882 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
883 ------------------------------------------------------------------------------
887 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
888 now defaults to force_rport. It is very important that phones requiring nat=no be
889 specifically set as such instead of relying on the default setting. If at all
890 possible, all devices should have nat settings configured in the general section as
891 opposed to configuring nat per-device.
892 * Added preferred_codec_only option in sip.conf. This feature limits the joint
893 codecs sent in response to an INVITE to the single most preferred codec.
894 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
895 to be used for the outgoing call. It must be one of the codecs configured
897 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
898 to be used for holding a private key. If tlsprivatekey is not specified,
899 tlscertfile is searched for both public and private key.
900 * Added tlsclientmethod option to sip.conf. This allows the protocol for
901 outbound client connections to be specified.
902 * The sendrpid parameter has been expanded to include the options
903 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
904 header to be sent (equivalent to setting sendrpid=yes) and setting
905 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
906 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
907 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
908 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
909 will accept the SDP even if the SDP version number is not properly incremented,
910 but will generate a warning in the log indicating that the SIP peer that sent
911 the SDP should have the 'ignoresdpversion' option set.
912 * The 'nat' option has now been been changed to have yes, no, force_rport, and
913 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
914 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
915 remote side requests it and disables symmetric RTP support. Setting it to
916 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
917 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
918 and enables symmetric RTP support.
919 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
920 response. This permits the master channel to know how each channel dialled
921 in a multi-channel setup resolved in an individual way. This carries a
922 performance penalty and can be disabled in sip.conf using the
923 'storesipcause' option.
924 * Added 'externtcpport' and 'externtlsport' options to allow custom port
925 configuration for the externip and externhost options when tcp or tls is used.
926 * Added support for message body (stored in content variable) to SIP NOTIFY message
927 accessible via AMI and CLI.
928 * Added 'media_address' configuration option which can be used to explicitly specify
929 the IP address to use in the SDP for media (audio, video, and text) streams.
930 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
931 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
933 * Added 'use_q850_reason' configuration option for generating and parsing
934 if available Reason: Q.850;cause=<cause code> header. It is implemented
935 in some gateways for better passing PRI/SS7 cause codes via SIP.
936 * When dialing SIP peers, a new component may be added to the end of the dialstring
937 to indicate that a specific remote IP address or host should be used when dialing
938 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
939 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
940 ability to selectively force bridged channels to also be encrypted is also
941 implemented. Branching in the dialplan can be done based on whether or not
942 a channel has secure media and/or signaling.
943 * Added directmediapermit/directmediadeny to limit which peers can send direct media
945 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
946 Charge messages to snom phones.
947 * Added support for G.719 media streams.
948 * Added support for 16khz signed linear media streams.
949 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
950 RTP has been outfitted with the same abilities.
951 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
952 available in device configurations as well as in the dial plan.
953 * Addition of the 'subscribe_network_change' option for turning on and off
954 res_stun_monitor module support in chan_sip.
955 * Addition of the 'auth_options_requests' option for turning on and off
956 authentication for OPTIONS requests in chan_sip.
960 * Add #tryinclude statement for config files. This provides the same
961 functionality as the #include statement however an asterisk module will
962 still load if the filename does not exist. Using the #include statement
963 Asterisk will not allow the module to load.
967 * Added rtsavesysname option into iax.conf to allow the systname to be saved
969 * Added the ability for chan_iax2 to inform the dialplan whether or not
970 encryption is being used. This interoperates with the SIP SRTP implementation
971 so that a secure SIP call can be bridged to a secure IAX call when the
972 dialplan requires bridged channels to be "secure".
973 * Addition of the 'subscribe_network_change' option for turning on and off
974 res_stun_monitor module support in chan_iax.
979 * Added ability to preset channel variables on indicated lines with the setvar
980 configuration option. Also, clearvars=all resets the list of variables back
982 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
983 See configs/res_pktccops.conf for more information.
985 XMPP Google Talk/Jingle changes
986 -------------------------------
987 * Added the externip option to gtalk.conf.
988 * Added the stunaddr option to gtalk.conf which allows for the automatic
989 retrieval of the external ip from a stun server.
993 * Added 'p' option to PickupChan() to allow for picking up channel by the first
994 match to a partial channel name.
995 * Added .m3u support for Mp3Player application.
996 * Added progress option to the app_dial D() option. When progress DTMF is
997 present, those values are sent immediately upon receiving a PROGRESS message
998 regardless if the call has been answered or not.
999 * Added functionality to the app_dial F() option to continue with execution
1000 at the current location when no parameters are provided.
1001 * Added the 'a' option to app_dial to answer the calling channel before any
1002 announcements or macros are executed.
1003 * Modified app_dial to set answertime when the called channel answers even if
1004 the called channel hangs up during playback of an announcement.
1005 * Modified app_dial 'r' option to support an additional parameter to play an
1006 indication tone from indications.conf
1007 * Added c() option to app_chanspy. This option allows custom DTMF to be set
1008 to cycle through the next available channel. By default this is still '*'.
1009 * Added x() option to app_chanspy. This option allows DTMF to be set to
1010 exit the application.
1011 * The Voicemail application has been improved to automatically ignore messages
1012 that only contain silence.
1013 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
1014 associated mailbox(es) to be greetings-only.
1015 * The ChanSpy application now has the 'S' option, which makes the application
1016 automatically exit once it hits a point where no more channels are available
1018 * The ChanSpy application also now has the 'E' option, which spies on a single
1019 channel and exits when that channel hangs up.
1020 * The MeetMe application now turns on the DENOISE() function by default, for
1021 each participant. In our tests, this has significantly decreased background
1022 noise (especially noisy data centers).
1023 * Voicemail now permits storage of secrets in a separate file, located in the
1024 spool directory of each individual user. The control for this is located in
1025 the "passwordlocation" option in voicemail.conf. Please see the sample
1026 configuration for more information.
1027 * The ChanIsAvail application now exposes the returned cause code using a separate
1028 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
1029 * Added 'd' option to app_followme. This option disables the "Please hold"
1031 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
1032 received will terminate recording.
1033 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
1034 Previously the folder could only be set per context, but has now been extended
1035 using the imapfolder option.
1036 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
1037 * Voicemail now allows the pager date format to be specified separately from the
1039 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
1040 to allow joining, leaving, and sending text to group chats.
1041 * MeetMe has a new option 'G' to play an announcement before joining a conference.
1042 * Page has a new option 'A(x)' which will playback an announcement simultaneously
1043 to all paged phones (and optionally excluding the caller's one using the new
1044 option 'n') before the call is bridged.
1045 * The 'f' option to Dial has been augmented to take an optional argument. If no
1046 argument is provided, the 'f' option works as it always has. If an argument is
1047 provided, then the connected party information of all outgoing channels created
1048 during the Dial will be set to the argument passed to the 'f' option.
1049 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
1051 * The OSP lookup application adds in/outbound network ID, optional security,
1052 number portability, QoS reporting, destination IP port, custom info and service
1054 * Added new application VMSayName that will play the recorded name of the voicemail
1055 user if it exists, otherwise will play the mailbox number.
1056 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
1057 retrieve state for a particular bridge, where <name> is the conference name
1058 * app_directory now allows exiting at any time using the operator or pound key.
1059 * Voicemail now supports setting a locale per-mailbox.
1060 * Two new applications are provided for declining counting phrases in multiple
1061 languages. See the application notes for SayCountedNoun and SayCountedAdj for
1063 * Voicemail now runs the externnotify script when pollmailboxes is activated and
1065 * Voicemail now includes rdnis within msgXXXX.txt file.
1066 * ExternalIVR now supports IPv6 addresses.
1067 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
1068 at https://wiki.asterisk.org/wiki/x/oQBB
1069 * ParkedCall and Park can now specify the parking lot to use.
1073 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
1074 over SRV records associated with a specific service. From the CLI, type
1075 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
1076 details on how these may be used.
1077 * PITCH_SHIFT dialplan function added. This function can be used to modify the
1078 pitch of a channel's tx and rx audio streams.
1079 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
1080 setting various connected line and redirecting party information.
1081 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
1082 support ISDN subaddressing.
1083 * The CHANNEL() function now supports the "name" and "checkhangup" options.
1084 * For DAHDI channels, the CHANNEL() dialplan function now allows
1085 the dialplan to request changes in the configuration of the active
1086 echo canceller on the channel (if any), for the current call only.
1089 exten => s,n,Set(CHANNEL(echocan_mode)=off)
1091 The possible values are:
1093 on - normal mode (the echo canceller is actually reinitialized)
1095 fax - FAX/data mode (NLP disabled if possible, otherwise completely
1097 voice - voice mode (returns from FAX mode, reverting the changes that
1098 were made when FAX mode was requested)
1099 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
1100 and setting variables on the channel which created the current channel.
1101 Administrators should take care to avoid naming conflicts, when multiple
1102 channels are dialled at once, especially when used with the Local channel
1103 construct (which all could set variables on the master channel). Usage
1104 of the HASH() dialplan function, with the key set to the name of the slave
1105 channel, is one approach that will avoid conflicts.
1106 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
1108 * func_odbc now allows multiple row results to be retrieved without using
1109 mode=multirow. If rowlimit is set, then additional rows may be retrieved
1110 from the same query by using the name of the function which retrieved the
1111 first row as an argument to ODBC_FETCH().
1112 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
1113 dialplan. This function returns the content of the received message.
1114 * Added REPLACE, which searches a given variable name for a set of characters,
1115 then either replaces them with a single character or deletes them.
1116 * Added PASSTHRU, which literally passes the same argument back as its return
1117 value. The intent is to be able to use a literal string argument to
1118 functions that currently require a variable name as an argument.
1119 * HASH-associated variables now can be inherited across channel creation, by
1120 prefixing the name of the hash at assignment with the appropriate number of
1121 underscores, just like variables.
1122 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
1123 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
1124 whether or not channels that are bridged to the current channel will be
1125 required to have secure signaling and/or media.
1126 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
1127 the current channel has secure signaling and/or media.
1128 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
1129 "no_media_path" option.
1130 Returns "0" if there is a B channel associated with the call.
1131 Returns "1" if no B channel is associated with the call. The call is either
1132 on hold or is a call waiting call.
1133 * Added option to dialplan function CDR(), the 'f' option
1134 allows for high resolution times for billsec and duration fields.
1135 * FILE() now supports line-mode and writing.
1136 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
1137 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
1141 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
1142 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
1143 and is set when a dynamic feature is triggered.
1144 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
1145 to dynamically create a new parking lot matching the value this varible is
1147 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
1148 features.conf that should be the base for dynamic parkinglots.
1149 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
1150 parkinglot should have.
1151 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
1152 parkinglot should have.
1153 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
1158 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
1159 timeout has expired.
1160 * Added 'R' option to app_queue. This option stops moh and indicates ringing
1161 to the caller when an Agent's phone is ringing. This can be used to indicate
1162 to the caller that their call is about to be picked up, which is nice when
1163 one has been on hold for an extened period of time.
1164 * A new config option, penaltymemberslimit, has been added to queues.conf.
1165 When set this option will disregard penalty settings when a queue has too
1167 * A new option, 'I' has been added to both app_queue and app_dial.
1168 By setting this option, Asterisk will not update the caller with
1169 connected line changes or redirecting party changes when they occur.
1170 * A 'relative-periodic-announce' option has been added to queues.conf. When
1171 enabled, this option will cause periodic announce times to be calculated
1172 from the end of announcements rather than from the beginning.
1173 * The autopause option in queues.conf can be passed a new value, "all." The
1174 result is that if a member becomes auto-paused, he will be paused in all
1175 queues for which he is a member, not just the queue that failed to reach
1177 * Added dialplan function QUEUE_EXISTS to check if a queue exists
1178 * The queue logger now allows events to optionally propagate to a file,
1179 even when realtime logging is turned on. Additionally, realtime logging
1180 supports sending the event arguments to 5 individual fields, although it
1181 will fallback to the previous data definition, if the new table layout is
1184 mISDN channel driver (chan_misdn) changes
1185 ----------------------------------------
1186 * Added display_connected parameter to misdn.conf to put a display string
1187 in the CONNECT message containing the connected name and/or number if
1188 the presentation setting permits it.
1189 * Added display_setup parameter to misdn.conf to put a display string
1190 in the SETUP message containing the caller name and/or number if the
1191 presentation setting permits it.
1192 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
1193 indicate the dialplan settings are to be obtained from the asterisk
1195 * Made misdn.conf parameter callerid accept the "name" <number> format
1196 used by the rest of the system.
1197 * Made use the nationalprefix and internationalprefix misdn.conf
1198 parameters to prefix any received number from the ISDN link if that
1199 number has the corresponding Type-Of-Number. NOTE: This includes
1200 comparing the incoming call's dialed number against the MSN list.
1201 * Added the following new parameters: unknownprefix, netspecificprefix,
1202 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
1203 received number from the ISDN link if that number has the corresponding
1205 * Added new dialplan application misdn_command which permits controlling
1206 the CCBS/CCNR functionality.
1207 * Added new dialplan function mISDN_CC which permits retrieval of various
1208 values from an active call completion record.
1209 * For PTP, you should manually send the COLR of the redirected-to party
1210 for an incomming redirected call if the incoming call could experience
1211 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
1212 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
1213 if the REDIRECTING(from-num) is not empty.
1214 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
1215 option on all of the REDIRECTING statements before dialing the
1216 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
1217 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
1218 redirecting-to presentation (COLR) when it becomes available.
1219 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
1222 thirdparty mISDN enhancements
1223 -----------------------------
1224 mISDN has been modified by Digium, Inc. to greatly expand facility message
1226 * Enhanced COLP support for call diversion and transfer.
1227 * CCBS/CCNR support.
1229 The latest modified mISDN v1.1.x based version is available at:
1230 http://svn.digium.com/svn/thirdparty/mISDN/trunk
1231 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
1233 Tagged versions of the modified mISDN code are available under:
1234 http://svn.digium.com/svn/thirdparty/mISDN/tags
1235 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
1237 libpri channel driver (chan_dahdi) DAHDI changes
1238 -------------------------------------------
1239 * The channel variable PRIREDIRECTREASON is now just a status variable
1240 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
1241 to read and alter the reason.
1242 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
1243 redirected-to party for an incomming redirected call if the incoming call
1244 could experience further redirects. Just set the
1245 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
1246 to the COLR. A call has been redirected if the REDIRECTING(count) is not
1248 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
1249 use the inhibit(i) option on all of the REDIRECTING statements before
1250 dialing the redirected-to party. You still have to set the
1251 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
1252 will update the redirecting-to presentation (COLR) when it becomes available.
1253 * Added the ability to ignore calls that are not in a Multiple Subscriber
1254 Number (MSN) list for PTMP CPE interfaces.
1255 * Added dynamic range compression support for dahdi channels. It is
1256 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
1257 * Added support for ISDN calling and called subaddress with partial support
1258 for connected line subaddress.
1259 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
1260 * Added handling of received HOLD/RETRIEVE messages and the optional ability
1261 to transfer a held call on disconnect similar to an analog phone.
1262 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
1263 Will reroute/deflect an outgoing call when receive the message.
1264 Can use the DAHDISendCallreroutingFacility to send the message for the
1266 * Added standard location to add options to chan_dahdi dialing:
1267 Dial(DAHDI/g1[/extension[/options]])
1270 R Reverse charging indication
1271 * Added Reverse Charging Indication (Collect calls) send/receive option.
1272 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
1273 Dial(DAHDI/g1/extension/R)
1274 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
1275 (requires latest LibPRI)
1276 * Added ability to send/receive keypad digits in the SETUP message.
1277 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
1278 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
1279 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
1280 (requires latest LibPRI)
1281 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
1282 to eliminate tromboned calls. A tromboned call goes out an interface and comes
1283 back into the same interface. Tromboned calls happen because of call routing,
1284 call deflection, call forwarding, and call transfer.
1285 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
1286 * Added the ability to support call waiting calls. (The SETUP has no B channel
1288 * Added Malicious Call ID (MCID) event to the AMI call event class.
1289 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
1291 Asterisk Manager Interface
1292 --------------------------
1293 * The Hangup action now accepts a Cause header which may be used to
1294 set the channel's hangup cause.
1295 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
1296 to specify a separate .pem file to hold a private key. By default sslcert
1297 is used to hold both the public and private key.
1298 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
1299 for options containing the 'tls' prefix. For example, 'sslenable' is now
1300 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
1301 across all .conf files. All affected sample.conf files have been modified to
1302 reflect this change. Previous options such as 'sslenable' still work,
1303 but options with the 'tls' prefix are preferred.
1304 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
1305 in a channel. (res_mutestream.so)
1306 * The configuration file manager.conf now supports a channelvars option, which
1307 specifies a list of channel variables to include in each channel-oriented
1309 * The redirect command now has new parameters ExtraContext, ExtraExtension,
1310 and ExtraPriority to allow redirecting the second channel to a different
1311 location than the first.
1312 * Added new event "JabberStatus" in the Jabber module to monitor buddies
1314 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
1315 in a MixMonitor recording.
1316 * The 'iax2 show peers' output is now similar to the expected output of
1318 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
1320 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
1321 AOC-E messages on a channel.
1322 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
1323 conform more closely to similar events.
1324 * Added a new eventfilter option per user to allow whitelisting and blacklisting
1326 * Added optional parkinglot variable for park command.
1327 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
1328 if CallerIDNum and CallerIDName headers are also present.
1330 Channel Event Logging
1331 ---------------------
1332 * A new interface, CEL, is introduced here. CEL logs single events, much like
1333 the AMI, but it differs from the AMI in that it logs to db backends much
1334 like CDR does; is based on the event subsystem introduced by Russell, and
1335 can share in all its benefits; allows multiple backends to operate like CDR;
1336 is specialized to event data that would be of concern to billing sytems,
1337 like CDR. Backends for logging and accounting calls have been produced,
1338 but a new CDR backend is still in development.
1342 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
1343 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
1344 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
1345 * Multiple files and formats can now be specified in cdr_custom.conf.
1346 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
1347 See configs/cdr_syslog.conf.sample for more information.
1348 * A 'sequence' field has been added to CDRs which can be combined with
1349 linkedid or uniqueid to uniquely identify a CDR.
1350 * Handling of billsec and duration field has changed. If your table definition
1351 specifies those fields as float,double or similar they will now be logged with
1352 microsecond accuracy instead of a whole integer.
1354 Calendaring for Asterisk
1355 ------------------------
1356 * A new set of modules were added supporing calendar integration with Asterisk.
1357 Dialplan functions for reading from and writing to calendars are included,
1358 as well as the ability to execute dialplan logic upon calendar event notifications.
1359 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
1360 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
1361 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
1362 2003 support does not support forms-based authentication).
1364 Call Completion Supplementary Services for Asterisk
1365 ---------------------------------------------------
1366 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
1367 DAHDI/ISDN supports call completion for the following switch types:
1368 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
1369 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
1371 Multicast RTP Support
1372 ---------------------
1373 * A new RTP engine and channel driver have been added which supports Multicast RTP.
1374 The channel driver can be used with the Page application to perform multicast RTP
1375 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
1376 Type can be either basic or linksys.
1377 Destination is the IP address and port for the RTP packets.
1378 Control address is specific to the linksys type and is used for sending the control
1379 packets unique to them.
1381 Security Events Framework
1382 -------------------------
1383 * Asterisk has a new C API for reporting security events. The module res_security_log
1384 sends these events to the "security" logger level. Currently, AMI is the only
1385 Asterisk component that reports security events. However, SIP support will be
1386 coming soon. For more information on the security events framework, see the
1387 "Asterisk Security Framework" section of the Asterisk wiki at
1388 https://wiki.asterisk.org/wiki/x/wgBQ
1389 * SIP support was added in Asterisk 10
1390 * This API now supports IPv6 addresses
1394 * A technology independent fax frontend (res_fax) has been added to Asterisk.
1395 * A spandsp based fax backend (res_fax_spandsp) has been added.
1396 * The app_fax module has been deprecated in favor of the res_fax module and
1397 the new res_fax_spandsp backend.
1398 * The SendFAX and ReceiveFAX applications now send their log messages to a
1399 'fax' logger level, instead of to the generic logger levels. To see these
1400 messages, the system's logger.conf file will need to direct the 'fax' logger
1401 level to one or more destinations; the logger.conf.sample file includes an
1402 example of how to do this. Note that if the 'fax' logger level is *not*
1403 directed to at least one destination, log messages generated by these
1404 applications will be lost, and that if the 'fax' logger level is directed to
1405 the console, the 'core set verbose' and 'core set debug' CLI commands will
1406 have no effect on whether the messages appear on the console or not.
1410 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
1411 Now, in order to enable transmitting silence during record the transmit_silence
1412 option should be used. transmit_silence_during_record remains a valid option, but
1413 defaults to the behavior of the transmit_silence option.
1414 * Addition of the Unit Test Framework API for managing registration and execution
1415 of unit tests with the purpose of verifying the operation of C functions.
1416 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
1417 XMPP text messages to the remote JID.
1418 * Modules.conf has a new option - "require" - that marks a module as critical for
1419 the execution of Asterisk.
1420 If one of the required modules fail to load, Asterisk will exit with a return
1422 * An 'X' option has been added to the asterisk application which enables #exec support.
1423 This allows #exec to be used in asterisk.conf.
1424 * jabber.conf supports a new option auth_policy that toggles auto user registration.
1425 * A new lockconfdir option has been added to asterisk.conf to protect the
1426 configuration directory (/etc/asterisk by default) during reloads.
1427 * The parkeddynamic option has been added to features.conf to enable the creation
1428 of dynamic parkinglots.
1429 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
1430 the reportalarms config option.
1431 * chan_dahdi supports dialing configuring and dialing by device file name.
1432 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
1433 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
1434 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
1435 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
1436 Handy for the above name-based syntax as it does not depend on
1437 initialization order.
1438 * The Realtime dialplan switch now caches entries for 1 second. This provides a
1439 significant increase in performance (about 3X) for installations using this switchtype.
1440 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
1441 AIS. For more information, please see the Distributed Device State section of the
1442 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1443 * The addition of G.719 pass-through support.
1444 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
1445 during device configuration.
1446 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
1447 have less than 3 lines on the LCD.
1448 * Realtime now supports database failover. See the sample extconfig.conf for details.
1449 * The addition of improved translation path building for wideband codecs. Sample
1450 rate changes during translation are now avoided unless absolutely necessary.
1451 * The addition of the res_stun_monitor module for monitoring and reacting to network
1452 changes while behind a NAT.
1456 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
1457 optionally accept a filename, to apply the setting only to the code generated from
1458 that source file when Asterisk was built. However, there are some modules in Asterisk
1459 that are composed of multiple source files, so this did not result in the behavior
1460 that users expected. In this version, 'core set debug' and 'core set verbose'
1461 can optionally accept *module* names instead (with or without the .so extension),
1462 which applies the setting to the entire module specified, regardless of which source
1463 files it was built from.
1464 * New 'manager show settings' command showing the current settings loaded from
1466 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
1467 the channel hangup request to all channels.
1468 * Added a "core reload" CLI command that executes a global reload of Asterisk.
1470 ------------------------------------------------------------------------------
1471 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
1472 ------------------------------------------------------------------------------
1476 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
1477 Snom phones use this for call pickup of extensions that the phone is
1479 * Added support for setting the domain in the URI for caller of an
1480 outbound call by using the SIPFROMDOMAIN channel variable.
1481 * Added a new configuration option "remotesecret" for authentication to
1482 remote services. For backwards compatibility, "secret" still has the
1483 same function as before, but now you can configure both a remote secret and a
1484 local secret for mutual authentication.
1485 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
1486 the sound will be played to the target of an attended transfer
1487 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
1488 finer control over how many peers Asterisk will qualify and the gap between them
1489 when all peers need to be qualified at the same time.
1490 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
1491 (either globally or for a specific peer), chan_sip will treat any SDP data
1492 it receives as new data and update the media stream accordingly. By
1493 default, Asterisk will only modify the media stream if the SDP session
1494 version received is different from the current SDP session version. This
1495 option is required to interoperate with devices that have non-standard SDP
1496 session version implementations (observed with Microsoft OCS). This option
1497 is disabled by default.
1498 * The parsing of register => lines in sip.conf has been modified to allow a port
1499 to be present in the "user" portion. Please see the sip.conf.sample file for more
1501 * Added support for subscribing to MWI on a remote server and making the status available
1502 as a mailbox. Please see the sip.conf.sample file for more information.
1503 * Added a function to remove SIP headers added in the dialplan before the
1504 first INVITE is generated - SIPRemoveHeader()
1505 * Channel variables set with setvar= in a device configuration is now
1506 set both for inbound and outbound calls.
1507 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
1511 * Added immediate option to iax.conf
1512 * Added forceencryption option to iax.conf
1513 * Added Encryption and Trunk status to manager command "iaxpeers"
1517 * The configuration file now holds separate sections for devices and lines.
1518 Please have a look at configs/skinny.conf.sample and change your skinny.conf
1523 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
1524 support for LibOpenR2. http://www.libopenr2.org/
1525 * The UK option waitfordialtone has been added for use with BT analog
1527 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
1528 is used in conjunction with the 'faxdetect' configuration option. When
1529 'faxbuffers' is used and fax tones are detected, the channel will dynamically
1530 switch to the configured faxbuffers policy. For example, to use 6 buffers
1531 and a 'full' buffer policy for a fax transmission, add:
1533 The faxbuffers configuration will be in affect until the call is torn down.
1534 * Added service message support for 4ESS/5ESS switches.
1538 * For DAHDI channels, the CHANNEL() dialplan function now
1539 supports changing the channel's buffer policy (for the current
1540 call only), using this syntax:
1542 exten => s,n,Set(CHANNEL(buffers)=6,full)
1544 This would change the channel to the 'full' buffer policy and
1545 6 (six) buffers. Possible options for this setting are the same
1546 as those in chan_dahdi.conf.
1547 * Added a new dialplan function, CURLOPT, which permits setting various
1548 options that may be useful with the CURL dialplan function, such as
1549 cookies, proxies, connection timeouts, passwords, etc.
1550 * Permit the syntax and synopsis fields of the corresponding dialplan
1551 functions to be individually set from func_odbc.conf.
1552 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
1553 * func_odbc now may specify an insert query to execute, when the write query
1554 affects 0 rows (usually indicating that no such row exists).
1555 * Added a new dialplan function, LISTFILTER, which permits removing elements
1556 from a set list, by name. Uses the same general syntax as the existing CUT
1557 and FIELDQTY dialplan functions, which also manage lists.
1558 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
1559 obtaining realtime data from the dialplan.
1560 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
1561 a subroutine when using the GoSub() and Return() applications.
1562 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
1563 of "core show function AUDIOHOOK_INHERIT" from the CLI
1564 * Added AES_ENCRYPT. For information on its use, please see the output
1565 of "core show function AES_ENCRYPT" from the CLI
1566 * Added AES_DECRYPT. For information on its use, please see the output
1567 of "core show function AES_DECRYPT" from the CLI
1568 * func_odbc now supports database transactions across multiple queries.
1572 * Scheduled meetme conferences may now have their end times extended by
1574 * app_authenticate now gives the ability to select a prompt other than
1576 * app_directory now pays attention to the searchcontexts setting in
1577 voicemail.conf and will look through all contexts, if no context is
1578 specified in the initial argument.
1579 * A new application, Originate, has been introduced, that allows asynchronous
1580 call origination from the dialplan.
1581 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
1582 in addition to the setting in the "general" context.
1583 * Added ConfBridge dialplan application which does conference bridges without
1584 DAHDI. For information on its use, please see the output of
1585 "core show application ConfBridge" from the CLI.
1589 * The Asterisk CLI has a new command, "channel redirect", which is similar in
1590 operation to the AMI Redirect action.
1591 * extensions.conf now allows you to use keyword "same" to define an extension
1592 without actually specifying an extension. It uses exactly the same pattern
1593 as previously used on the last "exten" line. For example:
1594 exten => 123,1,NoOp(something)
1595 same => n,SomethingElse()
1596 * musiconhold.conf classes of type 'files' can now use relative directory paths,
1597 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
1598 * All deprecated CLI commands are removed from the sourcecode. They are now handled
1599 by the new clialiases module. See cli_aliases.conf.sample file.
1600 * Times within timespecs are now accurate down to the minute. This is a change
1601 from historical Asterisk, which only provided timespecs rounded to the nearest
1602 even (read: evenly divisible by 2) minute mark.
1603 * The realtime switch now supports an option flag, 'p', which disables searches for
1605 * In addition to a time range and date range, timespecs now accept a 5th optional
1606 argument, timezone. This allows you to perform time checks on alternate
1607 timezones, especially if those daylight savings time ranges vary from your
1608 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
1610 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
1611 give you the correct output for an asterisk box behind nat. It will give you the
1612 externhost and localnet settings.
1613 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
1614 can connect calls in passthrough mode, as well as record and play back files.
1615 * Successful and unsuccessful call pickup can now be alerted through sounds, by
1616 using pickupsound and pickupfailsound in features.conf.
1617 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
1618 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
1619 instead of the /var/run/asterisk.pid where it used to be. This will make
1620 installs as non-root easier to manage.
1625 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
1626 be written; they will no longer be explicitly written.
1628 Asterisk Manager Interface
1629 --------------------------
1630 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
1631 a non-empty value) in your request. If you do this, any pending AMI events will
1632 *not* be included in the response to your request as they would normally, but
1633 will be left in the event queue for the next request you make to retrieve. For
1634 some applications, this will allow you to guarantee that you will only see
1635 events in responses to 'WaitEvent' actions, and can better know when to expect them.
1636 To know whether the Asterisk server supports this header or not, your client can
1637 inspect the first response back from the server to see if it includes this header:
1639 Pragma: SuppressEvents
1641 If this is included, the server supports event suppression.
1643 * Added 4 new Actions to list skinny device(s) and line(s)
1649 LDAP Schema File Additions
1650 --------------------------
1651 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
1652 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
1654 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
1655 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
1656 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
1657 * Removed redundant IPaddr (there's already IPAddress)
1658 - Gives more configuration Flags for SIP-Users available (tested)
1659 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
1660 without extensibleObject (which really should be the last resort); gives
1661 also additional possibilities for LDAP-filter
1663 ------------------------------------------------------------------------------
1664 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
1665 ------------------------------------------------------------------------------
1667 Device State Handling
1668 ---------------------
1669 * The event infrastructure in Asterisk got another big update to help support
1670 distributed events. It currently supports distributed device state and
1671 distributed Voicemail MWI (Message Waiting Indication). A new module has
1672 been merged, res_ais, which facilitates communicating events between servers.
1673 It uses the SAForum AIS (Service Availability Forum Application Interface
1674 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
1675 a cluster of Asterisk servers, and to share events between them. For more
1676 information on setting this up, refer to the Distributed Device State section
1677 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1681 * Added a new dialplan function, AST_CONFIG(), which allows you to access
1682 variables from an Asterisk configuration file.
1683 * The JACK_HOOK function now has a c() option to supply a custom client name.
1684 * Added two new dialplan functions from libspeex for audio gain control and
1685 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
1686 rx directions of a channel from the dialplan.
1687 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
1688 based on other parameters. The default is still to search based on the
1689 forwarding station ID. However, there are new options that allow you to search
1690 based on the message desk terminal ID, or the message desk number.
1691 * TIMEOUT() has been modified to be accurate down to the millisecond.
1692 * ENUM*() functions now include the following new options:
1693 - 'u' returns the full URI and does not strip off the URI-scheme.
1694 - 's' triggers ISN specific rewriting
1695 - 'i' looks for branches into an Infrastructure ENUM tree
1696 - 'd' for a direct DNS lookup without any flipping of digits.
1697 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
1698 * CHANNEL() now has options for the maximum, minimum, and standard or normal
1699 deviation of jitter, rtt, and loss for a call using chan_sip.
1701 DAHDI channel driver (chan_dahdi) Changes
1702 ----------------------------------------
1703 * Channels can now be configured using named sections in chan_dahdi.conf, just
1704 like other channel drivers, including the use of templates.
1705 * The default for pridialplan has changed from 'national' to 'unknown'.
1709 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
1710 to something that matches the pattern a hint will be created using the contents
1711 and variables evaluated.
1712 * Dialplan matching has been extended to allow an extension to return to the
1713 PBX core to wait for more digits. This is done by using the new dialplan
1714 application called "Incomplete". This will permit a whole new level of
1715 extension control, by giving the administrator more control over early
1716 matches employing one of the short-circuit pattern match operators. Note
1717 that custom applications can trigger this same behavior by returning the
1718 special value AST_PBX_INCOMPLETE.
1722 * Directory now permits both first and last names to be matched at the same
1723 time. In addition, the number of digits to enter of the name can be set in
1724 the arguments to Directory; previously, you could enter only 3, regardless
1725 of how many names are in your company. For large companies, this should be
1727 * Voicemail now permits a mailbox setting to wrap around from first to last
1728 messages, if the "messagewrap" option is set to a true value.
1729 * Voicemail now permits an external script to be run, for password validation.
1730 The script should output "VALID" or "INVALID" on stdout, depending upon the
1731 wish to validate or invalidate the password given. Arguments are:
1732 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
1734 * Dial has a new option: F(context^extension^pri), which permits a callee to
1735 continue in the dialplan, at the specified label, if the caller hangs up.
1736 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
1737 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
1738 * The Jack application now has a c() option to supply a custom client name.
1739 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
1740 like the pre-existing whisper mode, except that the spy can also talk to the
1741 participant on the bridged channel as well.
1742 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
1743 to be spoken instead of the channel name or number. For more information on the
1744 use of this option, issue the command "core show application ChanSpy" from the
1746 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
1747 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
1748 words, if using the 'd' option, it is not possible to enter a number to append to
1749 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
1750 change to whisper mode, and pressing 6 will change to barge mode.
1751 * ExternalIVR now takes several options that affect the way it performs, as
1752 well as having several new commands. Please see the External IVR page on the Asterisk
1753 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
1754 * Added ability to communicate over a TCP socket instead of forking a child process for the
1755 ExternalIVR application.
1756 * ChanIsAvail has a new option, 'a', which will return all available channels instead
1757 of just the first one if you give the function more then one channel to check.
1758 * PrivacyManager now takes an option where you can specify a context where the
1759 given number will be matched. This way you have more control over who is allowed
1760 and it stops the people who blindly enter 10 digits.
1761 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
1762 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
1763 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
1764 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
1765 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
1766 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
1767 * The Dial() application no longer copies the language used by the caller to the callee's
1768 channel. If you desire for the caller's channel's language to be used for file playback
1769 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
1770 * SendImage() no longer hangs up the channel on error; instead, it sets the
1771 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
1772 'UNSUPPORTED'. This change makes SendImage() more consistent with other
1774 * Park has a new option, 's', which silences the announcement of the parking space number.
1775 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
1776 invalid input and will be assumed to mean that no timeout is desired.
1780 * Added DNS manager support to registrations for peers referencing peer entries.
1781 DNS manager runs in the background which allows DNS lookups to be run asynchronously
1782 as well as periodically updating the IP address. These properties allow for
1783 better performance as well as recovery in the event of an IP change.
1784 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
1785 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
1786 These changes also provide performance improvements for call setup and tear down.
1787 * Added ability to specify registration expiry time on a per registration basis in
1789 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
1791 * Added t38pt_usertpsource option. See sip.conf.sample for details.
1792 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
1793 * 'sip show peers' and 'sip show users' display their entries sorted in
1794 alphabetical order, as opposed to the order they were in, in the config
1796 * Videosupport now supports an additional option, "always", which always sets
1797 up video RTP ports, even on clients that don't support it. This helps with
1798 callfiles and certain transfers to ensure that if two video phones are
1799 connected, they will always share video feeds.
1803 * Existing DNS manager lookups extended to check for SRV records.
1804 * IAX2 encryption support has been improved to support periodic key rotation
1805 within a call for enhanced security. The option "keyrotate" has been
1806 provided to disable this functionality to preserve backwards compatibility
1807 with older versions of IAX2 that do not support key rotation.
1811 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
1812 data tree based on the given <path>.
1813 * New CLI command "data show providers" that will display all the registered
1815 * New CLI command, "config reload <file.conf>" which reloads any module that
1816 references that particular configuration file. Also added "config list"
1817 which shows which configuration files are in use.
1818 * New CLI commands, "pri show version" and "ss7 show version" that will
1819 display which version of libpri and libss7 are being used, respectively.
1820 A new API call was added so trunk will now have to be compiled against
1821 a versions of libpri and libss7 that have them or it will not know that
1822 these libraries exist.
1823 * The commands "core show globals", "core set global" and "core set chanvar" has
1824 been deprecated in favor of the more semanticly correct "dialplan show globals",
1825 "dialplan set chanvar" and "dialplan set global".
1826 * New CLI command "dialplan show chanvar" to list all variables associated
1827 with a given channel.
1831 * Addresses managed by DNS manager now can check to see if there is a DNS
1832 SRV record for a given domain and will use that hostname/port if present.
1834 AMI - The manager (TCP/TLS/HTTP)
1835 --------------------------------
1836 * The Status command now takes an optional list of variables to display
1837 along with channel status.
1838 * The QueueEntry event now also includes the channel's uniqueid
1842 * res_odbc no longer has a limit of 1023 total possible unshared connections,
1843 as some people were running into this limit. This limit has been increased
1848 * The TRANSFER queue log entry now includes the the caller's original
1849 position in the transferred-from queue.
1850 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
1851 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
1852 as well as an explanation about timeout options in general
1853 * Added a new option - C - for forcing the "answered elsewhere" flag on
1854 cancellation of calls in to members of the queue. This is to avoid the
1855 call to a member of a queue having the call listed as a "missed call".
1859 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
1860 adaptive capabilities. What this means in practical terms is that if your
1861 realtime table lacks critical fields, Asterisk will now emit warnings to
1862 that effect. Also, some of the realtime drivers have the ability (if
1863 configured) to automatically add those columns to the table with the
1864 correct type and length.
1868 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
1869 the 'setvar' option to cause a given audio file to be played upon completion
1870 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
1871 Skinny channels only.
1872 * You can now compile Asterisk against the Hoard Memory Allocator, see the
1873 Hoard page on the Asterisk wiki for more information:
1874 https://wiki.asterisk.org/wiki/x/pQBB
1875 * Config file variables may now be appended to, by using the '+=' append
1876 operator. This is most helpful when working with long SQL queries in
1877 func_odbc.conf, as the queries no longer need to be specified on a single
1879 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
1880 which will add a second to the billsec when the ending
1881 time is set, if the number in the microseconds field of the end time is
1882 greater than the number of microseconds in the answer time. This allows
1883 users to count the 'initiated' seconds in their billing records.
1885 ------------------------------------------------------------------------------
1886 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
1887 ------------------------------------------------------------------------------
1889 AMI - The manager (TCP/TLS/HTTP)
1890 --------------------------------
1891 * Manager has undergone a lot of changes, all of them documented
1892 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
1893 * Manager version has changed to 1.1
1894 * Added a new action 'CoreShowChannels' to list currently defined channels
1895 and some information about them.
1896 * Added a new action 'SIPshowregistry' to list SIP registrations.
1897 * Added TLS support for the manager interface and HTTP server
1898 * Added the URI redirect option for the built-in HTTP server
1899 * The output of CallerID in Manager events is now more consistent.
1900 CallerIDNum is used for number and CallerIDName for name.
1901 * Enable https support for builtin web server.
1902 See configs/http.conf.sample for details.
1903 * Added a new action, GetConfigJSON, which can return the contents of an
1904 Asterisk configuration file in JSON format. This is intended to help
1905 improve the performance of AJAX applications using the manager interface
1907 * SIP and IAX manager events now use "ChannelType" in all cases where we
1908 indicate channel driver. Previously, we used a mixture of "Channel"
1909 and "ChannelDriver" headers.
1910 * Added a "Bridge" action which allows you to bridge any two channels that
1911 are currently active on the system.
1912 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
1913 the voicemail users setup.
1914 * Added 'DBDel' and 'DBDelTree' manager commands.
1915 * cdr_manager now reports events via the "cdr" level, separating it from
1916 the very verbose "call" level.
1917 * Manager users are now stored in memory. If you change the manager account
1918 list (delete or add accounts) you need to reload manager.
1919 * Added Masquerade manager event for when a masquerade happens between
1921 * Added "manager reload" command for the CLI
1922 * Lots of commands that only provided information are now allowed under the
1923 Reporting privilege, instead of only under Call or System.
1924 * The IAX* commands now require either System or Reporting privilege, to
1925 mirror the privileges of the SIP* commands.
1926 * Added ability to retrieve list of categories in a config file.
1927 * Added ability to retrieve the content of a particular category.
1928 * Added ability to empty a context.
1929 * Created new action to create a new file.
1930 * Updated delete action to allow deletion by line number with respect to category.
1931 * Added new action insert to add new variable to category at specified line.
1932 * Updated action newcat to allow new category to be inserted in file above another
1934 * Added new event "JitterBufStats" in the IAX2 channel
1935 * Originate now requires the Originate privilege and, if you want to call out
1936 to a subshell, it requires the System privilege, as well. This was done to
1937 enhance manager security.
1938 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
1939 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
1940 or manager show command Atxfer from the CLI
1941 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
1942 details or manager show command IAXregistry from the CLI
1946 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
1947 state in the dialplan, as well as creating custom device states that are
1948 controllable from the dialplan.
1949 * Extend CALLERID() function with "pres" and "ton" parameters to
1950 fetch string representation of calling number presentation indicator
1951 and numeric representation of type of calling number value.
1952 * MailboxExists converted to dialplan function
1953 * A new option to Dial() for telling IP phones not to count the call
1954 as "missed" when dial times out and cancels.
1955 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
1956 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
1957 held for any given channel. Also, locks are automatically freed when a
1959 * Added HINT() dialplan function that allows retrieving hint information.
1960 Hints are mappings between extensions and devices for the sake of
1961 determining the state of an extension. This function can retrieve the list
1962 of devices or the name associated with a hint.
1963 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
1965 * Added SYSINFO() dialplan function which allows retrieval of system information
1966 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
1967 the existence of a dialplan target.
1968 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
1969 upper and lower case, respectively.
1970 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
1971 ID for the call (not the Asterisk call ID or unique ID), provided that the
1972 channel driver supports this. For SIP, you get the SIP call-ID for the
1973 bridged channel which you can store in the CDR with a custom field.
1977 * Added CLI permissions, config file: cli_permissions.conf
1978 default is to allow all commands for every local user/group.
1979 Also this new feature added three new CLI commands:
1980 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
1981 - cli reload permissions
1982 - cli show permissions
1983 * New CLI command "core show hint" (usage: core show hint <exten>)
1984 * New CLI command "core show settings"
1985 * Added 'core show channels count' CLI command.
1986 * Added the ability to set the core debug and verbose values on a per-file basis.
1987 * Added 'queue pause member' and 'queue unpause member' CLI commands
1988 * Ability to set process limits ("ulimit") without restarting Asterisk
1989 * Enhanced "agi debug" to print the channel name as a prefix to the debug
1990 output to make debugging on busy systems much easier.
1991 * New CLI commands "dialplan set extenpatternmatching true/false"
1992 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
1993 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
1994 listed in the startup_commands section of cli.conf will get executed.
1995 * Added a CLI command, "devstate change", which allows you to set custom device
1996 states from the func_devstate module that provides the DEVICE_STATE() function
1997 and handling of the "Custom:" devices.
1998 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
1999 sorted into the different possible callbacks, with the number of entries
2000 currently scheduled for each. Gives you a feel for how busy the sip channel
2002 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
2003 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
2004 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
2008 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
2009 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
2010 for a received call. If it is detected, the channel will jump to the
2011 'fax' extension in the dialplan.
2012 * The default SIP useragent= identifier now includes the Asterisk version
2013 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
2014 If set, and the incoming request carries authentication info,
2015 the username to match in the users list is taken from the Digest header
2016 rather than from the From: field. This feature is considered experimental.
2017 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
2018 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
2019 * The "localmask" setting was removed in version 1.2 and the reminder about it
2020 being removed is now also removed.
2021 * A new option "busylevel" for setting a level of calls where asterisk reports
2022 a device as busy, to separate it from call-limit. This value is also added
2023 to the SIP_PEER dialplan function.
2024 * A new realtime family called "sipregs" is now supported to store SIP registration
2025 data. If this family is defined, "sippeers" will be used for configuration and
2026 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
2027 registration data, as before.
2028 * The SIPPEER function have new options for port address, call and pickup groups
2029 * Added support for T.140 realtime text in SIP/RTP
2030 * The "checkmwi" option has been removed from sip.conf, as it is no longer
2031 required due to the restructuring of how MWI is handled. See the descriptions
2032 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
2033 for more information.
2034 * Added rtpdest option to CHANNEL() dialplan function.
2035 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
2036 * SIP now adds a header to the CANCEL if the call was answered by another phone
2037 in the same dial command, or if the new c option in dial() is used.
2038 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
2039 states it is not needed. For phones, however, that do require it the "registertrying" option
2040 has been added so it can be enabled.
2041 * A new option called "callcounter" (global/peer/user level) enables call counters needed
2042 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
2043 used to enable this functionality).
2044 * New settings for timer T1 and timer B on a global level or per device. This makes it
2045 possible to force timeout faster on non-responsive SIP servers. These settings are
2046 considered advanced, so don't use them unless you have a problem.
2047 * Added a dial string option to be able to set the To: header in an INVITE to any
2049 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
2050 the qualify frequency.
2051 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
2052 were not properly torn down due to network or endpoint failures during an established
2054 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
2055 and configs/sip.conf.sample for more information on how it is used.
2056 * Added a new configuration option "authfailureevents" that enables manager events when
2057 a peer can't authenticate properly.
2058 * Added DNS manager support to registrations for peers not referencing a peer entry.
2062 * Added the trunkmaxsize configuration option to chan_iax2.
2063 * Added the srvlookup option to iax.conf
2064 * Added support for OSP. The token is set and retrieved through the CHANNEL()
2067 XMPP Google Talk/Jingle changes
2068 -------------------------------
2069 * Added the bindaddr option to gtalk.conf.
2073 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
2074 * Proper codec support in chan_skinny.
2075 * Added settings for IP and Ethernet QoS requests
2079 * Added separate settings for media QoS in mgcp.conf
2081 Console Channel Driver changes
2082 ------------------------------
2083 * Added experimental support for video send & receive to chan_oss.
2084 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
2087 Phone channel changes (chan_phone)
2088 ----------------------------------
2089 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
2091 H.323 channel Changes
2092 ---------------------
2093 * H323 remote hold notification support added (by NOTIFY message
2094 and/or H.450 supplementary service)
2096 Local channel changes
2097 ---------------------
2098 * The device state functionality in the Local channel driver has been updated
2099 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
2100 to just UNKNOWN if the extension exists.
2101 * Added jitterbuffer support for chan_local. This allows you to use the
2102 generic jitterbuffer on incoming calls going to Asterisk applications.
2103 For example, this would allow you to use a jitterbuffer for an incoming
2104 SIP call to Voicemail by putting a Local channel in the middle. This
2105 feature is enabled by using the 'j' option in the Dial string to the Local
2106 channel in conjunction with the existing 'n' option for local channels.
2107 * A 'b' option has been added which causes chan_local to return the actual channel
2108 that is behind it when queried. This is useful for transfer scenarios as the
2109 actual channel will be transferred, not the Local channel.
2111 Agent channel changes
2112 ----------------------
2113 * The ackcall and endcall options are now supplemented with options acceptdtmf
2114 and enddtmf. These allow for the DTMF keypress to be configurable. The options
2115 default to their old hard-coded values ('#' and '*' respectively) so this should
2116 not break any existing agent installations.
2118 DAHDI channel driver (chan_dahdi) Changes
2119 ----------------------------------------
2120 * SS7 support (via libss7 library)
2121 * In India, some carriers transmit CID via dtmf. Some code has been added
2122 that will handle some situations. The cidstart=polarity_IN choice has been added for
2123 those carriers that transmit CID via dtmf after a polarity change.
2124 * CID matching information is now shown when doing 'dialplan show'.
2125 * Added dahdi show version CLI command.
2126 * Added setvar support to chan_dahdi.conf channel entries.
2127 * Added two new options: mwimonitor and mwimonitornotify. These options allow
2128 you to enable MWI monitoring on FXO lines. When the MWI state changes,
2129 the script specified in the mwimonitornotify option is executed. An internal
2130 event indicating the new state of the mailbox is also generated, so that
2131 the normal MWI facilities in Asterisk work as usual.
2132 * Added signalling type 'auto', which attempts to use the same signalling type
2133 for a channel as configured in DAHDI. This is primarily designed for analog
2134 ports, but will also work for digital ports that are configured for FXS or FXO
2135 signalling types. This mode is also the default now, so if your chan_dahdi.conf
2136 does not specify signalling for a channel (which is unlikely as the sample
2137 configuration file has always recommended specifying it for every channel) then
2138 the 'auto' mode will be used for that channel if possible.
2139 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
2140 state for a channel; also ensured that the DNDState Manager event is
2141 emitted no matter how the DND state is set or cleared.
2145 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
2146 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
2147 for details. This new channel driver allows you to use Nortel i2002,
2148 i2004, and i2050 phones with Asterisk.
2149 * Added a new channel driver, chan_console, which uses portaudio as a cross
2150 platform audio interface. It was written as a channel driver that would
2151 work with Mac CoreAudio, but portaudio supports a number of other audio
2152 interfaces, as well. Note that this channel driver requires v19 or higher
2153 of portaudio; older versions have a different API.
2157 * Added the ability to specify arguments to the Dial application when using
2158 the DUNDi switch in the dialplan.
2159 * Added the ability to set weights for responses dynamically. This can be
2160 done using a global variable or a dialplan function. Using the SHELL()
2161 function would allow you to have an external script set the weight for
2163 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
2164 functions will allow you to initiate a DUNDi query from the dialplan,
2165 find out how many results there are, and access each one.
2166 * Added the ability to specifiy a port for a dundi peer.
2170 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
2171 functions will allow you to initiate an ENUM lookup from the dialplan,
2172 and Asterisk will cache the results. ENUMRESULT can be used to access
2173 the results without doing multiple DNS queries.
2177 * Added the ability to customize which sound files are used for some of the
2178 prompts within the Voicemail application by changing them in voicemail.conf
2179 * Added the ability for the "voicemail show users" CLI command to show users
2180 configured by the dynamic realtime configuration method.
2181 * MWI (Message Waiting Indication) handling has been significantly
2182 restructured internally to Asterisk. It is now totally event based
2183 instead of polling based. The voicemail application will notify other
2184 modules that have subscribed to MWI events when something in the mailbox
2186 This also means that if any other entity outside of Asterisk is changing
2187 the contents of mailboxes, then the voicemail application still needs to
2188 poll for changes. Examples of situations that would require this option
2189 are web interfaces to voicemail or an email client in the case of using
2190 IMAP storage. So, two new options have been added to voicemail.conf
2191 to account for this: "pollmailboxes" and "pollfreq". See the sample
2192 configuration file for details.
2193 * Added "tw" language support
2194 * Added support for storage of greetings using an IMAP server
2195 * Added ability to customize forward, reverse, stop, and pause keys for message playback
2196 * SMDI is now enabled in voicemail using the smdienable option.
2197 * A "lockmode" option has been added to asterisk.conf to configure the file
2198 locking method used for voicemail, and potentially other things in the
2199 future. The default is the old behavior, lockfile. However, there is a
2200 new method, "flock", that uses a different method for situations where the
2201 lockfile will not work, such as on SMB/CIFS mounts.
2202 * Added the ability to backup deleted messages, to ease recovery in the case
2203 that a user accidentally deletes a message, and discovers that they need it.
2204 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
2205 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
2206 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
2207 voicemail boxes. The SMDI interface can also poll for MWI changes when some
2208 outside entity is modifying the state of the mailbox (such as IMAP storage or
2209 a web interface of some kind).
2210 * Added the support for marking messages as "urgent." There are two methods to accomplish
2211 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
2212 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
2213 the message as urgent after he has recorded a voicemail by following the voice instructions.
2214 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
2219 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
2220 used across multiple queues.
2221 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
2222 setqueueentryvar options for each queue, see queues.conf.sample for details.
2223 * Added keepstats option to queues.conf which will keep queue
2224 statistics during a reload.
2225 * setinterfacevar option in queues.conf also now sets a variable
2226 called MEMBERNAME which contains the member's name.
2227 * Added 'Strategy' field to manager event QueueParams which represents
2228 the queue strategy in use.
2229 * Added option to run macro when a queue member is connected to a caller,
2230 see queues.conf.sample for details.
2231 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
2232 does not count paused queue members as unavailable.
2233 * Added min-announce-frequency option to queues.conf which allows you to control the
2234 minimum amount of time between queue announcements for use when the caller's queue
2235 position changes frequently.
2236 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
2238 * Added ability for non-realtime queues to have realtime members
2239 * Added the "linear" strategy to queues.
2240 * Added the "wrandom" strategy to queues.
2241 * Added new channel variable QUEUE_MIN_PENALTY
2242 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
2243 rules in queuerules.conf. See configs/queuerules.conf.sample for details
2244 * Added a new parameter for member definition, called state_interface. This may be
2245 used so that a member may be called via one interface but have a different interface's
2246 device state reported.
2247 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
2248 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
2249 "manager show command QueueReset."
2250 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
2251 specified by the periodic-announce option, then one will be chosen randomly when it is time
2252 to play a periodic announcment
2253 * New configuration options: announce-position now takes two more values in addition to "yes" and
2254 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
2255 announce-position-limit. By setting announce-position to "limit" callers will only have their
2256 position announced if their position is less than what is specified by announce-position-limit.
2257 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
2258 will be told that their are more than announce-position-limit callers waiting.
2259 * Two new queue log events have been added. An ADDMEMBER event will be logged
2260 when a realtime queue member is added and a REMOVEMEMBER event will be logged
2261 when a realtime queue member is removed. Since there is no calling channel associated
2262 with these events, the string "REALTIME" is placed where the channel's unique id
2263 is typically placed.
2264 * The configuration method for the "joinempty" and "leavewhenempty" options has
2265 changed to a comma-separated list of methods of determining member availability
2266 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
2267 values are still accepted for backwards-compatibility, though.
2268 * The average talktime is now calculated on queues. This information is reported via the
2269 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
2270 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
2275 * The 'o' option to provide an optimization has been removed and its functionality
2276 has been enabled by default.
2277 * When a conference is created, the UNIQUEID of the channel that caused it to be
2278 created is stored. Then, every channel that joins the conference will have the
2279 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
2280 callers that come and go from long standing conferences.
2281 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
2282 except it does operations on a channel by name, instead of number in a conference.
2283 This is a very useful feature in combination with the 'X' option to ChanSpy.
2284 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
2286 * Added new RealTime functionality to provide support for scheduled conferencing.
2287 This includes optional messages to the caller if they attempt to join before
2288 the schedule start time, or to allow the caller to join the conference early.
2289 Also included is optional support for limiting the number of callers per
2290 RealTime conference.
2291 * Added the S() and L() options to the MeetMe application. These are pretty
2292 much identical to the S() and L() options to Dial(). They let you set
2293 timeouts for the conference, as well as have warning sounds played to
2294 let the caller know how much time is left, and when it is running out.
2295 * Added the ability to do "meetme concise" with the "meetme" CLI command.
2296 This extends the concise capabilities of this CLI command to include
2297 listing all conferences, instead of an addition to the other sub commands
2298 for the "meetme" command.
2299 * Added the ability to specify the music on hold class used to play into the
2300 conference when there is only one member and the M option is used.
2301 * Added MEETME_INFO dialplan function which provides a way to query
2302 various properties of a Meetme conference.
2303 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
2304 and *84: record in-conf
2306 Other Dialplan Application Changes
2307 ----------------------------------
2308 * Argument support for Gosub application
2309 * From the to-do lists: straighten out the app timeout args:
2310 Wait() app now really does 0.3 seconds- was truncating arg to an int.
2311 WaitExten() same as Wait().
2312 Congestion() - Now takes floating pt. argument.
2313 Busy() - now takes floating pt. argument.
2314 Read() - timeout now can be floating pt.
2315 WaitForRing() now takes floating pt timeout arg.
2316 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
2317 * Added 's' option to Page application.
2318 * Added an optional timeout argument to the Page application.
2319 * Added 'E', 'V', and 'P' commands to ExternalIVR.
2320 * Added 'o' and 'X' options to Chanspy.
2321 * Added a new dialplan application, Bridge, which allows you to bridge the
2322 calling channel to any other active channel on the system.
2323 * Added the ability to specify a music on hold class to play instead of ringing
2324 for the SLATrunk application.
2325 * The Read application no longer exits the dialplan on error. Instead, it sets
2326 READSTATUS to ERROR, which you can catch and handle separately.
2327 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
2328 of asking for verification of each name, one at a time.
2329 * Privacy() no longer uses privacy.conf, as all options are specifyable as
2330 direct options to the app.
2331 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
2333 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
2334 * The ChannelRedirect application no longer exits the dialplan if the given channel
2335 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
2336 or NOCHANNEL if the given channel was not found.
2337 * The silencethreshold setting that was previously configurable in multiple
2338 applications is now settable globally via dsp.conf.
2340 Music On Hold Changes
2341 ---------------------
2342 * A new option, "digit", has been added for music on hold classes in
2343 musiconhold.conf. If this is set for a music on hold class, a caller
2344 listening to music on hold can press this digit to switch to listening
2345 to this music on hold class.
2346 * Support for realtime music on hold has been added.
2347 * In conjunction with the realtime music on hold, a general section has
2348 been added to musiconhold.conf, its sole variable is cachertclasses. If this
2349 is set, then music on hold classes found in realtime will be cached in memory.
2353 * AEL upgraded to use the Gosub with Arguments instead
2354 of Macro application, to hopefully reduce the problems
2355 seen with the artificially low stack ceiling that
2356 Macro bumps into. Macros can only call other Macros
2357 to a depth of 7. Tests run using gosub, show depths
2358 limited only by virtual memory. A small test demonstrated
2359 recursive call depths of 100,000 without problems.
2360 -- in addition to this, all apps that allowed a macro
2361 to be called, as in Dial, queues, etc, are now allowing
2362 a gosub call in similar fashion.
2363 * AEL now generates LOCAL(argname) declarations when it
2364 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
2365 etc. That makes the arguments local in scope. The user
2366 can define their own local variables in macros, now,
2367 by saying "local myvar=someval;" or using Set() in this
2368 fashion: Set(LOCAL(myvar)=someval); ("local" is now
2370 * utils/conf2ael introduced. Will convert an extensions.conf
2371 file into extensions.ael. Very crude and unfinished, but
2372 will be improved as time goes by. Should be useful for a
2373 first pass at conversion.
2374 * aelparse will now read extensions.conf to see if a referenced
2375 macro or context is there before issueing a warning.
2376 * AEL parser sets a local channel variable ~~EXTEN~~, to
2377 preserve the value of ${EXTEN} thru switch statements.
2378 * New operator in $[...] expressions: the ~~ operator serves
2379 as a concatenation operator. AT THE MOMENT, it is really only
2380 necessary and useful in AEL, especially in if() expressions.
2381 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
2382 any enclosing double-quotes, and evaluate to the value of a
2383 concatenated with the value of b. For example if a is set to
2384 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
2385 evaluate to xyzabc .
2388 Call Features (res_features) Changes
2389 ------------------------------------
2390 * Added the parkedcalltransfers option to features.conf
2391 * Added parkedcallparking option to control one touch parking w/ parking
2393 * Added parkedcallhangup option to control disconnect feature w/ parking
2395 * Added parkedcallrecording option to control one-touch record w/ parking
2397 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
2398 parkedcalltransfers option support for multiple parking lots.
2399 * Added BRIDGE_FEATURES variable to set available features for a channel
2400 * The built-in method for doing attended transfers has been updated to
2401 include some new options that allow you to have the transferee sent
2402 back to the person that did the transfer if the transfer is not successful.
2403 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
2404 in features.conf.sample.
2405 * Added support for configuring named groups of custom call features in
2406 features.conf. This means that features can be written a single time, and
2407 then mapped into groups of features for different key mappings or easier
2409 * Updated the ParkedCall application to allow you to not specify a parking
2410 extension. If you don't specify a parking space to pick up, it will grab
2411 the first one available.
2412 * Added cli command 'features reload' to reload call features from features.conf
2413 * Moved into core asterisk binary.
2414 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
2415 * Added the ability for custom parking lots to be configured with their own
2416 parking extension with the parkext option.
2418 Language Support Changes
2419 ------------------------
2420 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
2421 * Added support for the Hungarian language for saying numbers, dates, and times.
2425 * Added SPEECH commands for speech recognition. A complete listing can be found
2427 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
2428 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
2429 does not behave as expected; the native command needs to be used, instead.
2430 * Added the ability to perform SRV lookups on fast AGI calls. To use this
2431 feature, simply use hagi: instead of agi: as the protocol portion
2432 of the URI parameter to the AGI function call in your dial plan. Also note
2433 that specifying a port number in the AGI URI will disable SRV lookups,
2434 even if you use the hagi: protocol.
2435 * No longer support MSG_OOB flag on HANGUP.
2439 * Added rotatestrategy option to logger.conf, along with two new options:
2440 "timestamp" which will use the time to name the logger files instead of
2441 sequence number; and "rotate", which rotates the names of the log files,
2442 similar to the way syslog rotates files.
2443 * Added exec_after_rotate option to logger.conf, which allows a system
2444 command to be run after rotation. This is primarily useful with
2445 rotatestrategy=rotate, to allow a limit on the number of log files kept
2446 and to ensure that the oldest log file gets deleted.
2447 * Added realtime support for the queue log
2451 * The cdr_manager module has a [mappings] feature, like cdr_custom,
2452 to add fields to the manager event from the CDR variables.
2453 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
2454 backend database CDR table. Specifically, additional, non-standard
2455 columns are supported, merely by setting the corresponding CDR variable in
2456 your dialplan. In addition, you may alias any column to another name (for
2457 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
2458 simply "alias src => ANI" in the configuration file). Records may be
2459 posted to more than one backend, simply by specifying multiple categories
2460 in the configuration file. And finally, you may filter which CDRs get
2461 posted to each backend, by specifying a filter (which the record must
2462 match) for the particular category. Filters are additive (meaning all
2463 rules must match to post that CDR).
2464 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
2465 module. Specifically, you may add additional columns into the table and
2466 they will be set, if you set the corresponding CDR variable name. Also,
2467 if you omit columns in your database table, they will be silently skipped
2468 (but a record will still be inserted, based on what columns remain). Note
2469 that the other two features from cdr_adaptive_odbc (alias and filter) are
2470 not currently supported.
2471 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
2472 has been disabled using the NoCDR application.
2474 Miscellaneous New Modules
2475 -------------------------
2476 * Added a new CDR module, cdr_sqlite3_custom.
2477 * Added a new realtime configuration module, res_config_sqlite
2478 * Added a new codec translation module, codec_resample, which re-samples
2479 signed linear audio between 8 kHz and 16 kHz to help support wideband
2481 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
2482 based on configuration templates that use Asterisk dialplan function and
2483 variable substitution. It should be possible to create phone profiles and
2484 templates that work for the majority of phones provisioned over http. It
2485 is currently only intended to provision a single user account per phone.
2486 An example profile and set of templates for Polycom phones is provided.
2487 NOTE: Polycom firmware is not included, but should be placed in
2488 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
2489 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
2490 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
2491 provided; there is a JACK() application, and a JACK_HOOK() function. Both
2492 interfaces create an input and output JACK port. The application makes
2493 these ports the endpoint of the call. The audio coming from the channel
2494 goes out the output port and whatever comes back in on the input port is
2495 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
2496 audiohook on the channel. This lets you run the audio coming from a
2497 channel through JACK, and whatever comes back in is what gets forwarded
2498 on as the channel's audio. This is very useful for building custom
2499 vocoders or doing recording or analysis of the channel's audio in another
2501 * Added a new module, res_config_curl, which permits using a HTTP POST url
2502 to retrieve, create, update, and delete realtime information from a remote
2503 web server. Note that this module requires func_curl.so to be loaded for
2504 backend functionality.
2505 * Added a new module, res_config_ldap, which permits the use of an LDAP
2506 server for realtime data access.
2507 * Added support for writing and running your dialplan in lua using the pbx_lua
2508 module. See configs/extensions.lua.sample for examples of how to do this.
2512 * Ability to use libcap to set high ToS bits when non-root
2513 on Linux. If configure is unable to find libcap then you
2514 can use --with-cap to specify the path.
2515 * Added maxfiles option to options section of asterisk.conf which allows you to specify
2516 what Asterisk should set as the maximum number of open files when it loads.
2517 * Added the jittertargetextra configuration option.
2518 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
2519 configuration files for the IP channel drivers. The new option is "cos".
2520 This information is also documented on the Asterisk wiki at
2521 https://wiki.asterisk.org/wiki/x/EYBG
2522 * When originating a call using AMI or pbx_spool that fails the reason for failure
2523 will now be available in the failed extension using the REASON dialplan variable.
2524 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
2525 It allows you to configure a prefix for auto-monitor recordings.
2526 * A new extension pattern matching algorithm, based on a trie, is introduced
2527 here, that could noticeably speed up mid-sized to large dialplans.
2528 It is NOT used by default, as duplicating the behaviour of the old pattern
2529 matcher is still under development. A config file option, in extensions.conf,
2530 in the [general] section, called "extenpatternmatchingnew", is by default
2531 set to false; setting that to true will force the use of the new algorithm.
2532 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
2533 be used to switch the algorithms at run time.
2534 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
2535 specifying which socket to use to connect to the running Asterisk daemon
2537 * Performance enhancements to the sched facility, which is used in
2538 the channel drivers, etc. Added hashtabs and doubly-linked lists
2539 to speed up deletion; start at the beginning or end of list to
2541 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
2542 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
2543 Added regression tests to the tests/ dir, also.
2544 * Added a refcount trace feature to astobj2 for those trying to balance
2545 object creation, deletion; work, play; space and time. See the
2546 notes in astobj2.h. Also, see utils/refcounter as well, as a
2547 quick way to find unbalanced refcounts in what could be a sea
2548 of objects that were balanced.
2549 * Added logging to 'make update' command. See update.log
2550 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
2551 do not come from the remote party.
2552 * Added the 'n' option to the SpeechBackground application to tell it to not
2553 answer the channel if it has not already been answered.
2554 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
2555 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
2557 * iLBC source code no longer included (see UPGRADE.txt for details)
2558 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
2559 deadlock is detected, a backtrace of the stack which led to the lock calls
2560 will be output to the CLI.
2561 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
2562 the "core show locks" CLI command will give lock information output as well
2563 as a backtrace of the stack which led to the lock calls.
2564 * users.conf now sports an optional alternateexts property, which permits
2565 allocation of additional extensions which will reach the specified user.
2566 * A new option for the configure script, --enable-internal-poll, has been added
2567 for use with systems which may have a buggy implementation of the poll system
2568 call. If you notice odd behavior such as the CLI being unresponsive on remote
2569 consoles, you may want to try using this option. This option is enabled by default
2570 on Darwin systems since it is known that the Darwin poll() implementation has
2574 --------------------
2575 * In addition to timing from DAHDI, there is a new timing module called
2576 res_timing_timerfd. In order to use this, you must be running Linux with
2577 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
2578 script will be able to tell if you have the requirements. From menuselect, select
2579 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob