1 ==============================================================================
3 === This file documents the new and/or enhanced functionality added in
4 === the Asterisk versions listed below. This file does NOT include
5 === changes in behavior that would not be backwards compatible with
6 === previous versions; for that information see the UPGRADE.txt file
7 === and the other UPGRADE files for older releases.
9 ==============================================================================
10 ------------------------------------------------------------------------------
11 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
12 ------------------------------------------------------------------------------
15 AMI (Asterisk Manager Interface)
17 * The SIPshowpeer action will now include a 'SubscribeContext' field for a peer
18 in its response if the peer has a subscribe context set.
20 * The SIPqualifypeer action now acknowledges the request once it has established
21 that the request is against a known peer. It also issues a new event,
22 'SIPqualifypeerdone', once the qualify action has been completed.
24 * The PlayDTMF action now supports an optional 'Duration' parameter. This
25 specifies the duration of the digit to be played, in milliseconds.
27 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
28 updates when changes occur instead of requiring the use of pollmailboxes.
30 * CLI Command 'Manager Show Commands' no longer truncates command names longer
31 than 15 characters and no longer shows authorization requirement for commands.
32 'Manager Show Command' now displays the privileges needed for using a given
33 manager command instead.
40 * Added general support for busy detection.
42 * Added ECAM command support for Sony Ericsson phones.
47 * The BRIDGE_FEATURES channel variable would previously only set features for
48 the calling party and would set this feature regardless of whether the
49 feature was in caps or in lowercase. Use of a caps feature for a letter
50 will now apply the feature to the calling party while use of a lowercase
51 letter will apply that feature to the called party.
53 * Add support for automixmonitor to the BRIDGE_FEATURES channel variable.
57 * When performing queue pause/unpause on an interface without specifying an
58 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
59 least one member of any queue exists for that interface.
63 * Add queue available hint. exten => 8501,hint,Queue:markq_avail
64 Note: the suffix '_avail' after the queuename.
65 Reports 'InUse' for no logged in agents or no free agents.
66 Reports 'Idle' when an agent is free.
70 * Redirecting reasons can now be set to arbitrary strings. This means
71 that the REDIRECTING dialplan function can be used to set the redirecting
72 reason to any string. It also allows for custom strings to be read as the
73 redirecting reason from SIP Diversion headers.
75 ------------------------------------------------------------------------------
76 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
77 ------------------------------------------------------------------------------
81 * The Asterisk build system will now build and install a shared library
82 (libasteriskssl.so) used to wrap various initialization and shutdown functions
83 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
84 that Asterisk can ensure that these functions do *not* get called by any
85 modules that are loaded into Asterisk, since they should only be called once
86 in any single process. If desired, this feature can be disabled by supplying
87 the "--disable-asteriskssl" option to the configure script.
89 * A new make target, 'full', has been added to the Makefile. This performs
90 the same compilation actions as make all, but will also scan the entirety of
91 each source file for documentation. This option is needed to generate AMI
92 event documentation. Note that your system must have Python in order for
93 this make target to succeed.
95 * The optimization portion of the build system has been reworked to avoid
96 broken builds on certain architectures. All architecture-specific
97 optimization has been removed in favor of using -march=native to allow gcc
98 to detect the environment in which it is running when possible. This can
99 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
101 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
102 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
104 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
105 previously parsed the header file to obtain the version of Asterisk, you
106 will now have to go through Asterisk to get the version information.
114 * Added 'F()' option. Similar to the dial option, this can be supplied with
115 arguments indicating where the callee should go after the caller is hung up,
116 or without options specified, the priority after the Queue will be used.
121 * Added menu action admin_toggle_mute_participants. This will mute / unmute
122 all non-admin participants on a conference. The confbridge configuration
123 file also allows for the default sounds played to all conference users when
124 this occurs to be overriden using sound_participants_unmuted and
125 sound_participants_muted.
127 * Added menu action participant_count. This will playback the number of
128 current participants in a conference.
130 * Added announcement configuration option to user profile. If set the sound
131 file will be played to the user, and only the user, upon joining the
137 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
138 channels respectively before the callee channels are called.
143 * Added support for IPv6.
145 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
146 external process will cause the current playlist to be cleared, including
147 stopping any audio file that is currently playing. This is useful when you
148 want to interrupt audio playback only when specific DTMF is entered by the
154 * A new option, 'I' has been added to app_followme. By setting this option,
155 Asterisk will not update the caller with connected line changes when they
156 occur. This is similar to app_dial and app_queue.
158 * The 'N' option is now ignored if the call is already answered.
160 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
161 and caller channels respectively before the callee channels are called.
163 * The winning FollowMe outgoing call is now put on hold if the caller put it on
169 * MixMonitor hooks now have IDs associated with them which can be used to
170 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
171 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
172 now accepts that ID as an argument.
174 * Added 'm' option, which stores a copy of the recording as a voicemail in the
180 * The connect action in app_mysql now allows you to specify a port number to
181 connect to. This is useful if you run a MySQL server on a non-standard
187 * Increased the default number of allowed destinations from 5 to 12.
192 * The app_page application now no longer depends on DAHDI or app_meetme. It
193 has been re-architected to use app_confbridge internally.
198 * Added queue options autopausebusy and autopauseunavail for automatically
199 pausing a queue member when their device reports busy or congestion.
201 * The 'ignorebusy' option for queue members has been deprecated in favor of
202 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
203 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
204 per interface basis. Individual ringinuse values can now be set in
205 queues.conf via an argument to member definitions. Lastly, the queue
206 'ringinuse' setting now only determines defaults for the per member
207 'ringinuse' setting and does not override per member settings like it does
210 * Added 'F()' option. Similar to the dial option, this can be supplied with
211 arguments indicating where the callee should go after the caller is hung up,
212 or without options specified, the priority after the Queue will be used.
214 * Added new option log_member_name_as_agent, which will cause the membername to
215 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
216 state_interface has been set.
218 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
222 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
223 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
224 changed arguments to SayUnixTime so that every option is truly optional even
225 when using multiple options (so that j option could be used without having to
226 manually specify timezone and format) There are other benefits, e.g., format
227 can now be used without specifying time zone as well.
232 * Addition of the VM_INFO function - see Function changes.
234 * The imapserver, imapport, and imapflags configuration options can now be
235 overriden on a user by user basis.
237 * When voicemail plays a message's envelope with saycid set to yes, when
238 reaching the caller id field it will play a recording of a file with the same
239 base name as the sender's callerid if there is a similarly named file in
240 <astspooldir>/recordings/callerids/
242 * Voicemails now contains a unique message identifier "msg_id", which is stored
243 in the message envelope with the sound files. IMAP backends will now store
244 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
245 backends will store the message identifier in a "msg_id" column. See
246 UPGRADE.txt for more information.
248 * Added VoiceMailPlayMsg application. This application will play a single
249 voicemail message from a mailbox. The result of the application, SUCCESS or
250 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
255 * Hangup handlers can be attached to channels using the CHANNEL() function.
256 Hangup handlers will run when the channel is hung up similar to the h
257 extension. The hangup_handler_push option will push a GoSub compatible
258 location in the dialplan onto the channel's hangup handler stack. The
259 hangup_handler_pop option will remove the last added location, and optionally
260 replace it with a new GoSub compatible location. The hangup_handler_wipe
261 option will remove all locations on the stack, and optionally add a new
264 * The expression parser now recognizes the ABS() absolute value function,
265 which will convert negative floating point values to positive values.
267 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
268 control of faxdetect.
270 * Addition of the VM_INFO function that can be used to retrieve voicemail
271 user information, such as the email address and full name.
272 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
275 * The REDIRECTING function now supports the redirecting original party id
278 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
279 lets you set some of the configuration options from the [general] section
280 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
281 the key sequence used to activate built-in features, such as blindxfer,
282 and automon. See the built-in documentation for details.
284 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
285 instead of simply the uri. This is the format that MessageSend() can use
286 in the from parameter for outgoing SIP messages.
288 * Added the PRESENCE_STATE function. This allows retrieving presence state
289 information from any presence state provider. It also allows setting
290 presence state information from a CustomPresence presence state provider.
291 See AMI/CLI changes for related commands.
293 * Added the AMI_CLIENT function to make manager account attributes available
294 to the dialplan. It currently supports returning the current number of
295 active sessions for a given account.
297 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
298 and the REDIRECTING functions.
306 * Added a manager event "LocalBridge" for local channel call bridges between
307 the two pseudo-channels created.
312 * Added dialtone_detect option for analog ports to disconnect incoming
313 calls when dialtone is detected.
315 * Added option colp_send to send ISDN connected line information. Allowed
316 settings are block, to not send any connected line information; connect, to
317 send connected line information on initial connect; and update, to send
318 information on any update during a call. Default is update.
320 * Add options namedcallgroup and namedpickupgroup to support installations
321 where a higher number of groups (>64) is required.
323 * Added support to use private party ID information with PRI calls.
328 * A new channel driver named chan_motif has been added which provides support for
329 Google Talk and Jingle in a single channel driver. This new channel driver includes
330 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
331 hold, unhold, and ringing notification. It is also compliant with the current Jingle
332 specification, current Google Jingle specification, and the original Google Talk
338 * Added NAT support for RTP. Setting in config is 'nat', which can be set
339 globally and overriden on a peer by peer basis.
341 * Direct media functionality has been added. Options in config are:
342 directmedia (directrtp) and directrtpsetup (earlydirect)
344 * ChannelUpdate events now contain a CallRef header.
349 * Asterisk will no longer substitute CID number for CID name in the display
350 name field if CID number exists without a CID name. This change improves
351 compatibility with certain device features such as Avaya IP500's directory
354 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
355 created using that setting to not be removed during SIP reload.
357 * Added settings recordonfeature and recordofffeature. When receiving an INFO
358 request with a "Record:" header, this will turn the requested feature on/off.
359 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
360 dynamic features must be enabled and configured properly on the requesting
361 channel for this to function properly.
363 * Add support to realtime for the 'callbackextension' option.
365 * When multiple peers exist with the same address, but differing
366 callbackextension options, incoming requests that are matched by address
367 will be matched to the peer with the matching callbackextension if it is
370 * Two new NAT options, auto_force_rport and auto_comedia, have been added
371 which set the force_rport and comedia options automatically if Asterisk
372 detects that an incoming SIP request crossed a NAT after being sent by
375 * NAT settings are now a combinable list of options. The equivalent of the
376 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
378 * Adds an option send_diversion which can be disabled to prevent
379 diversion headers from automatically being added to INVITE requests.
381 * Add support for lightweight NAT keepalive. If enabled a blank packet will
382 be sent to the remote host at a given interval to keep the NAT mapping open.
383 This can be enabled using the keepalive configuration option.
385 * Add option 'tonezone' to specify country code for indications. This option
386 can be set both globally and overridden for specific peers.
388 * The SIP Security Events Framework now supports IPv6.
390 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
391 between multiple user agents. When set, for directmedia reinvites,
392 Asterisk will not send an immediate reinvite on an incoming call leg. This
393 option is useful when peered with another SIP user agent that is known to
394 send immediate direct media reinvites upon call establishment.
396 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
399 * Add options subminexpiry and submaxexpiry to set limits of subscription
400 timer independently from registration timer settings. The setting of the
401 registration timer limits still is done by options minexpiry, maxexpiry
402 and defaultexpiry. For backwards compatibility the setting of minexpiry
403 and maxexpiry also is used to configure the subscription timer limits if
404 subminexpiry and submaxexpiry are not set in sip.conf.
406 * Set registration timer limits to default values when reloading sip
407 configuration and values are not set by configuration.
409 * Add options namedcallgroup and namedpickupgroup to support installations
410 where a higher number of groups (>64) is required.
412 * When a MESSAGE request is received, the address the request was received from
413 is now saved in the SIP_RECVADDR variable.
415 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
416 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
417 the ANI2/OLI information is set on the channel, which can be retrieved using
418 the CALLERID function.
420 * Peers can now be configured to support negotiation of ICE candidates using
421 the setting icesupport. See res_rtp_asterisk changes for more information.
423 * Added support for format attribute negotiation. See the Codecs changes for
426 * Extra headers specified with SIPAddHeader are sent with the REFER message
427 when using Transfer application. See refer_addheaders in sip.conf.sample.
429 * Added support to use private party ID information with calls.
434 * Added skinny version 17 protocol support.
439 * Added ability to use multiple lines for a single phone. This allows multiple
440 calls to occur on a single phone, using callwaiting and switching between calls.
442 * Added option 'sharpdial' allowing end dialing by pressing # key
444 * Added option 'interdigit_timer' to control phone dial timeout
446 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
448 * Added global 'debug' option, that enables debug in channel driver
450 * Added ability to translate on-screen menu in multiple languages. Tested on
451 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
452 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
455 * In addition to English added French and Russian languages for on-screen menus
457 * Reworked dialing number input: added dialing by timeout, immediate dial on
458 on dialplan compare, phone number length now not limited by screen size
460 * Added ability to pickup a call using features.conf defined value and
466 * Add options namedcallgroup and namedpickupgroup to support installations
467 where a higher number of groups (>64) is required.
469 * Added support to use private party ID information with calls.
474 * The minimum DTMF duration can now be configured in asterisk.conf
475 as "mindtmfduration". The default value is (as before) set to 80 ms.
476 (previously it was only available in source code)
478 * Named ACLs can now be specified in acl.conf and used in configurations that
479 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
480 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
481 working ACL. In addition, some CLI commands have been added to provide
482 show information and allow for module reloading - see CLI Changes.
484 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
485 items (separated by commas), and items in the rule can be negated by prefixing
486 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
487 longer necessray to control the order that the 'permit' and 'deny' columns are
488 returned from queries.
490 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
491 be used within the dynamic weight attribute when specifying a mapping.
493 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
494 header, instead of putting the user defined event name there. When enabled
495 the UserDefType header is added for user defined events. This feature is
496 enabled with the setting show_user_defined.
498 * Macro has been deprecated in favor of GoSub. For redirecting and connected
499 line purposes use the following variables instead of their macro equivalents:
500 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
501 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
502 cc_callback_macro in channel configurations.
504 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
507 * Call files now support the "early_media" option to connect with an outgoing
508 extension when early media is received.
510 * Added support to use private party ID information with calls.
515 * A new channel variable, AGIEXITONHANGUP, has been added which allows
516 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
517 AGI application would exit immediately after a channel hangup is detected.
519 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
520 are resolved and each address is attempted in turn until one succeeds or
524 AMI (Asterisk Manager Interface)
526 * The originate action now has an option "EarlyMedia" that enables the
527 call to bridge when we get early media in the call. Previously,
528 early media was disregarded always when originating calls using AMI.
530 * Added setvar= option to manager accounts (much like sip.conf)
532 * Originate now generates an error response if the extension given is not found
535 * MixMonitor will now show IDs associated with the mixmonitor upon creating
536 them if the i(variable) option is used. StopMixMonitor will accept
537 MixMonitorID as an option to close specific MixMonitors.
539 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
540 updated to include information about peers configured with
541 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
542 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
543 returned if auto_force_rport is not enabled.
545 * Added SIPpeerstatus manager command which will generate PeerStatus events
546 similar to the existing PeerStatus events found in chan_sip on demand.
548 * Hangup now can take a regular expression as the Channel option. If you want
549 to hangup multiple channels, use /regex/ as the Channel option. Existing
550 behavior to hanging up a single channel is unchanged, but if you pass a regex,
551 the manager will send you a list of channels back that were hung up.
553 * Support for IPv6 addresses has been added.
555 * AMI Events can now be documented in the Asterisk source. Note that AMI event
556 documentation is only generated when Asterisk is compiled using 'make full'.
557 See the CLI section for commands to display AMI event information.
559 * The AMI Hangup event now includes the AccountCode header so you can easily
560 correlate with AMI Newchannel events.
562 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
563 the StateInterface of the queue member.
565 * Added AMI event SessionTimeout in the Call category that is issued when a
566 call is terminated due to either RTP stream inactivity or SIP session timer
569 * CEL events can now contain a user defined header UserDefType. See core
570 changes for more information.
572 * OOH323 ChannelUpdate events now contain a CallRef header.
574 * Added PresenceState command. This command will report the presence state for
575 the given presence provider.
577 * Added Parkinglots command. This will list all parking lots as a series of
578 AMI Parkinglot events.
580 * Added MessageSend command. This behaves in the same manner as the
581 MessageSend application, and is a technolgoy agnostic mechanism to send out
582 of call text messages.
584 * Added "message" class authorization. This grants an account permission to
585 send out of call messages. Write-only.
590 * The "dialplan add include" command has been modified to create context a context
591 if one does not already exist. For instance, "dialplan add include foo into bar"
592 will create context "bar" if it does not already exist.
594 * A "dialplan remove context" command has been added to remove a context from
597 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
598 filenames of all running mixmonitors on a channel.
600 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
601 numeric instead of 0, 1, or 2.
603 * "stun show status" will show a table describing how the STUN client is
606 * "acl show [named acl]" will show information regarding a Named ACL. The
607 acl module can be reloaded with "reload acl".
609 * Added CLI command to display AMI event information - "manager show events",
610 which shows a list of all known and documented AMI events, and "manager show
611 event [event name]", which shows detail information about a specific AMI
614 * The result of the CLI command "queue show" now includes the state interface
615 information of the queue member.
617 * The command "core set verbose" will now set a separate level of logging for
618 each remote console without affecting any other console.
620 * Added command "cdr show pgsql status" to check connection status
622 * "sip show channel" will now display the complete route set.
624 * Added "presencestate list" command. This command will list all custom
625 presence states that have been set by using the PRESENCE_STATE dialplan
628 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
629 command. This changes a custom presence to a new state.
634 * Codec lists may now be modified by the '!' character, to allow succinct
635 specification of a list of codecs allowed and disallowed, without the
636 requirement to use two different keywords. For example, to specify all
637 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
639 * Add support for parsing SDP attributes, generating SDP attributes, and
640 passing it through. This support includes codecs such as H.263, H.264, SILK,
641 and CELT. You are able to set up a call and have attribute information pass.
642 This should help considerably with video calls.
644 * The iLBC codec can now use a system-provided iLBC library if one is installed,
645 just like the GSM codec.
649 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
650 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
654 * Asterisk version and build information is now logged at the beginning of a
657 * Threads belonging to a particular call are now linked with callids which get
658 added to any log messages produced by those threads. Log messages can now be
659 easily identified as involved with a certain call by looking at their call id.
660 Call ids may also be attached to log messages for just about any case where
661 it can be determined to be related to a particular call.
663 * Each logging destination and console now have an independent notion of the
664 current verbosity level. Logger.conf now allows an optional argument to
665 the 'verbose' specifier, indicating the level of verbosity sent to that
666 particular logging destination. Additionally, remote consoles now each
667 have their own verbosity level. The command 'core set verbose' will now set
668 a separate level for each remote console without affecting any other
674 * Added 'announcement' option which will play at the start of MOH and between
675 songs in modes of MOH that can detect transitions between songs (eg.
681 * New per parking lot options: comebackcontext and comebackdialtime. See
682 configs/features.conf.sample for more details.
684 * Channel variable PARKER is now set when comebacktoorigin is disabled in
687 * Channel variable PARKEDCALL is now set with the name of the parking lot
688 when a timeout occurs.
694 CDR Postgresql Driver
696 * Added command "cdr show pgsql status" to check connection status
699 CDR Adaptive ODBC Driver
701 * Added schema option for databases that support specifying a schema.
709 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
710 CALENDAR_WRITE has completed successfully.
715 * A new option, 'probation' has been added to rtp.conf
716 RTP in strictrtp mode can now require more than 1 packet to exit learning
717 mode with a new source (and by default requires 4). The probation option
718 allows the user to change the required number of packets in sequence to any
719 desired value. Use a value of 1 to essentially restore the old behavior.
720 Also, with strictrtp on, Asterisk will now drop all packets until learning
721 mode has successfully exited. These changes are based on how pjmedia handles
722 media sources and source changes.
724 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
725 enabled or disabled using the icesupport setting. A variety of other
726 settings have been introduced to configure STUN/TURN connections.
731 * A new module, res_corosync, has been introduced. This module uses the
732 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
733 of Asterisk servers to both Message Waiting Indication (MWI) and/or
734 Device State (presence) information. This module is very similar to, and
735 is a replacement for the res_ais module that was in previous releases of
741 * This module adds a cleaned up, drop-in replacement for res_jabber called
742 res_xmpp. This provides the same externally facing functionality but is
743 implemented differently internally. res_jabber has been deprecated in favor
744 of res_xmpp; please see the UPGRADE.txt file for more information.
749 * The safe_asterisk script has been updated to allow several of its parameters
750 to be set from environment variables. This also enables a custom run
751 directory of Asterisk to be specified, instead of defaulting to /tmp.
753 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
754 its value to determine the directory to assume is the top-level directory of
755 the source tree. If the variable is not set, it defaults to the current
756 behavior and uses the current working directory.
758 ------------------------------------------------------------------------------
759 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
760 ------------------------------------------------------------------------------
764 * Asterisk now has protocol independent support for processing text messages
765 outside of a call. Messages are routed through the Asterisk dialplan.
766 SIP MESSAGE and XMPP are currently supported. There are options in
767 jabber.conf and sip.conf to allow enabling these features.
768 -> jabber.conf: see the "sendtodialplan" and "context" options.
769 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
770 and "outofcall_message_context" options.
771 The MESSAGE() dialplan function and MessageSend() application have been
772 added to go along with this functionality. More detailed usage information
773 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
774 * If real-time text support (T.140) is negotiated, it will be preferred for
775 sending text via the SendText application. For example, via SIP, messages
776 that were once sent via the SIP MESSAGE request would be sent via RTP if
777 T.140 text is negotiated for a call.
781 * parkedmusicclass can now be set for non-default parking lots.
783 Asterisk Manager Interface
784 --------------------------
785 * PeerStatus now includes Address and Port.
786 * Added Hold events for when the remote party puts the call on and off hold
787 for chan_dahdi ISDN channels.
788 * Added new action MeetmeListRooms to list active conferences (shows same
789 data as "meetme list" at the CLI).
790 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
791 Description field that is set by 'description' in the channel configuration
793 * Added Uniqueid header to UserEvent.
794 * Added new action FilterAdd to control event filters for the current session.
795 This requires the system permission and uses the same filter syntax as
796 filters that can be defined in manager.conf
797 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
798 versions had some instances of the event converted, but others were left
799 as-is. All Unlink events should now be converted to Bridge events. The AMI
800 protocol version number was incremented to 1.2 as a result of this change.
803 --------------------------
804 * The HTTP Server can bind to IPv6 addresses.
807 --------------------------
808 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
809 with busydetect. usage example: busypattern=200,200,200,600
812 --------------------------
813 * New 'gtalk show settings' command showing the current settings loaded from
815 * The 'logger reload' command now supports an optional argument, specifying an
816 alternate configuration file to use.
817 * 'dialplan add extension' command will now automatically create a context if
818 the specified context does not exist with a message indicated it did so.
819 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
820 Description field which can be populated with 'description' in the channel
821 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
824 --------------------------
825 * The filter option in cdr_adaptive_odbc now supports negating the argument,
826 thus allowing records which do NOT match the specified filter.
827 * Added ability to log CONGESTION calls to CDR
830 --------------------------
831 * Ability to define custom SILK formats in codecs.conf.
832 * Addition of speex32 audio format with translation.
833 * CELT codec pass-through support and ability to define
834 custom CELT formats in codecs.conf.
835 * Ability to read raw signed linear files with sample rates
836 ranging from 8khz - 192khz. The new file extensions introduced
837 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
838 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
839 Skinny, H.323, etc) can still only support the following codecs:
840 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
841 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
842 Video: h261, h263, h263p, h264, mpeg4
847 --------------------------
848 * New highly optimized and customizable ConfBridge application capable of
849 mixing audio at sample rates ranging from 8khz-96khz.
850 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
851 and bridge profiles on a channel.
852 * CONFBRIDGE_INFO dialplan function capable of retrieving information
853 about a conference such as locked status and number of parties, admins,
855 * Addition of video_mode option in confbridge.conf for adding video support
856 into a bridge profile.
857 * Addition of the follow_talker video_mode in confbridge.conf. This video
858 mode dynamically switches the video feed to always display the loudest talker
859 supplying video in the conference.
863 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
864 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
865 variables from asterisk.conf.
869 * Addition of the JITTERBUFFER dialplan function. This function allows
870 for jitterbuffering to occur on the read side of a channel. By using
871 this function conference applications such as ConfBridge and MeetMe can
872 have the rx streams jitterbuffered before conference mixing occurs.
873 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
875 * Added STRREPLACE function. This function let's the user search a variable
876 for a given string to replace with another string as many times as the
877 user specifies or just throughout the whole string.
878 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
879 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
880 * Added extensions to chan_ooh323 in function CHANNEL()
882 libpri channel driver (chan_dahdi) DAHDI changes
883 --------------------------
884 * Added moh_signaling option to specify what to do when the channel's bridged
885 peer puts the ISDN channel on hold.
886 * Added display_send and display_receive options to control how the display ie
887 is handled. To send display text from the dialplan use the SendText()
888 application when the option is enabled.
889 * Added mcid_send option to allow sending a MCID request on a span.
892 --------------------------
893 * Added setvar option to calendar.conf to allow setting channel variables on
894 notification channels.
895 * Added "calendar show types" CLI command to list registered calendar
899 --------------------------
900 * Added two new options, r and t with file name arguments to record
901 single direction (unmixed) audio recording separate from the bidirectional
902 (mixed) recording. The mixed file name argument is optional now as long
903 as at least one recording option is used.
906 --------------------------
907 * Added a new option, l, which will disable local call optimization for
908 channels involved with the FollowMe thread. Use this option to improve
909 compatability for a FollowMe call with certain dialplan apps, options, and
913 --------------------------
914 * Added option "k" that will automatically close the conference when there's
915 only one person left when a user exits the conference.
918 --------------------------
919 * cel_pgsql now supports the 'extra' column for data added using the
920 CELGenUserEvent() application.
923 --------------------------
924 * Support for defining hints has been added to pbx_lua. See the 'hints' table
925 in the sample extensions.lua file for syntax details.
926 * Applications that perform jumps in the dialplan such as Goto will now
927 execute properly. When pbx_lua detects that the context, extension, or
928 priority we are executing on has changed it will immediately return control
929 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
930 the priority after the currently executing priority.
931 * An autoservice is now started by default for pbx_lua channels. It can be
932 stopped and restarted using the autoservice_stop() and autoservice_start()
936 --------------------------
937 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
938 into a FAXStatus event with an 'Operation' header that will be either
939 'send', 'receive', and 'gateway'.
940 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
941 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
942 feature will handle converting a fax call between an audio T.30 fax terminal
943 and an IFP T.38 fax terminal.
947 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
948 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
949 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
953 * Added general option negative_penalty_invalid default off. when set
954 members are seen as invalid/logged out when there penalty is negative.
955 for realtime members when set remove from queue will set penalty to -1.
956 * Added queue option autopausedelay when autopause is enabled it will be
957 delayed for this number of seconds since last successful call if there
958 was no prior call the agent will be autopaused immediately.
959 * Added member option ignorebusy this when set and ringinuse is not
960 will allow per member control of multiple calls as ringinuse does for
965 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
967 * Added 'k' option to MeetMe to automatically kill the conference when there's only
968 one participant left (much like a normal call bridge)
969 * Added extra argument to Originate to set timeout.
973 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
974 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
975 utility in the UTILS section of menuselect. If an existing astdb is found and no
976 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
977 convert an existing astdb to the SQLite3 version automatically at runtime.
981 * Modules marked as deprecated are no longer marked as building by default. Enabling
982 these modules is still available via menuselect.
986 * authdebug is now disabled by default. To enable this functionaility again
987 set authdebug = yes in iax.conf.
991 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
992 releases it was disabled.
996 * The PBX core previously made a call with a non-existing extension test for
997 extension s@default and jump there if the extension existed.
998 This was a bad default behaviour and violated the principle of least surprise.
999 It has therefore been changed in this release. It may affect some
1000 applications and configurations that rely on this behaviour. Most channel
1001 drivers have avoided this for many releases by testing whether the extension
1002 called exists before starting the PBX and generating a local error.
1003 This behaviour still exists and works as before.
1005 Extension "s" is used when no extension is given in a channel driver,
1006 like immediate answer in DAHDI or calling to a domain with no user part
1009 ------------------------------------------------------------------------------
1010 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
1011 ------------------------------------------------------------------------------
1015 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
1016 now defaults to force_rport. It is very important that phones requiring nat=no be
1017 specifically set as such instead of relying on the default setting. If at all
1018 possible, all devices should have nat settings configured in the general section as
1019 opposed to configuring nat per-device.
1020 * Added preferred_codec_only option in sip.conf. This feature limits the joint
1021 codecs sent in response to an INVITE to the single most preferred codec.
1022 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
1023 to be used for the outgoing call. It must be one of the codecs configured
1025 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
1026 to be used for holding a private key. If tlsprivatekey is not specified,
1027 tlscertfile is searched for both public and private key.
1028 * Added tlsclientmethod option to sip.conf. This allows the protocol for
1029 outbound client connections to be specified.
1030 * The sendrpid parameter has been expanded to include the options
1031 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
1032 header to be sent (equivalent to setting sendrpid=yes) and setting
1033 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
1034 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
1035 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
1036 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
1037 will accept the SDP even if the SDP version number is not properly incremented,
1038 but will generate a warning in the log indicating that the SIP peer that sent
1039 the SDP should have the 'ignoresdpversion' option set.
1040 * The 'nat' option has now been been changed to have yes, no, force_rport, and
1041 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
1042 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
1043 remote side requests it and disables symmetric RTP support. Setting it to
1044 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
1045 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
1046 and enables symmetric RTP support.
1047 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
1048 response. This permits the master channel to know how each channel dialled
1049 in a multi-channel setup resolved in an individual way. This carries a
1050 performance penalty and can be disabled in sip.conf using the
1051 'storesipcause' option.
1052 * Added 'externtcpport' and 'externtlsport' options to allow custom port
1053 configuration for the externip and externhost options when tcp or tls is used.
1054 * Added support for message body (stored in content variable) to SIP NOTIFY message
1055 accessible via AMI and CLI.
1056 * Added 'media_address' configuration option which can be used to explicitly specify
1057 the IP address to use in the SDP for media (audio, video, and text) streams.
1058 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
1059 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
1061 * Added 'use_q850_reason' configuration option for generating and parsing
1062 if available Reason: Q.850;cause=<cause code> header. It is implemented
1063 in some gateways for better passing PRI/SS7 cause codes via SIP.
1064 * When dialing SIP peers, a new component may be added to the end of the dialstring
1065 to indicate that a specific remote IP address or host should be used when dialing
1066 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
1067 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
1068 ability to selectively force bridged channels to also be encrypted is also
1069 implemented. Branching in the dialplan can be done based on whether or not
1070 a channel has secure media and/or signaling.
1071 * Added directmediapermit/directmediadeny to limit which peers can send direct media
1073 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
1074 Charge messages to snom phones.
1075 * Added support for G.719 media streams.
1076 * Added support for 16khz signed linear media streams.
1077 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
1078 RTP has been outfitted with the same abilities.
1079 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
1080 available in device configurations as well as in the dial plan.
1081 * Addition of the 'subscribe_network_change' option for turning on and off
1082 res_stun_monitor module support in chan_sip.
1083 * Addition of the 'auth_options_requests' option for turning on and off
1084 authentication for OPTIONS requests in chan_sip.
1088 * Add #tryinclude statement for config files. This provides the same
1089 functionality as the #include statement however an asterisk module will
1090 still load if the filename does not exist. Using the #include statement
1091 Asterisk will not allow the module to load.
1095 * Added rtsavesysname option into iax.conf to allow the systname to be saved
1096 on realtime updates.
1097 * Added the ability for chan_iax2 to inform the dialplan whether or not
1098 encryption is being used. This interoperates with the SIP SRTP implementation
1099 so that a secure SIP call can be bridged to a secure IAX call when the
1100 dialplan requires bridged channels to be "secure".
1101 * Addition of the 'subscribe_network_change' option for turning on and off
1102 res_stun_monitor module support in chan_iax.
1107 * Added ability to preset channel variables on indicated lines with the setvar
1108 configuration option. Also, clearvars=all resets the list of variables back
1110 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
1111 See configs/res_pktccops.conf for more information.
1113 XMPP Google Talk/Jingle changes
1114 -------------------------------
1115 * Added the externip option to gtalk.conf.
1116 * Added the stunaddr option to gtalk.conf which allows for the automatic
1117 retrieval of the external ip from a stun server.
1121 * Added 'p' option to PickupChan() to allow for picking up channel by the first
1122 match to a partial channel name.
1123 * Added .m3u support for Mp3Player application.
1124 * Added progress option to the app_dial D() option. When progress DTMF is
1125 present, those values are sent immediately upon receiving a PROGRESS message
1126 regardless if the call has been answered or not.
1127 * Added functionality to the app_dial F() option to continue with execution
1128 at the current location when no parameters are provided.
1129 * Added the 'a' option to app_dial to answer the calling channel before any
1130 announcements or macros are executed.
1131 * Modified app_dial to set answertime when the called channel answers even if
1132 the called channel hangs up during playback of an announcement.
1133 * Modified app_dial 'r' option to support an additional parameter to play an
1134 indication tone from indications.conf
1135 * Added c() option to app_chanspy. This option allows custom DTMF to be set
1136 to cycle through the next available channel. By default this is still '*'.
1137 * Added x() option to app_chanspy. This option allows DTMF to be set to
1138 exit the application.
1139 * The Voicemail application has been improved to automatically ignore messages
1140 that only contain silence.
1141 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
1142 associated mailbox(es) to be greetings-only.
1143 * The ChanSpy application now has the 'S' option, which makes the application
1144 automatically exit once it hits a point where no more channels are available
1146 * The ChanSpy application also now has the 'E' option, which spies on a single
1147 channel and exits when that channel hangs up.
1148 * The MeetMe application now turns on the DENOISE() function by default, for
1149 each participant. In our tests, this has significantly decreased background
1150 noise (especially noisy data centers).
1151 * Voicemail now permits storage of secrets in a separate file, located in the
1152 spool directory of each individual user. The control for this is located in
1153 the "passwordlocation" option in voicemail.conf. Please see the sample
1154 configuration for more information.
1155 * The ChanIsAvail application now exposes the returned cause code using a separate
1156 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
1157 * Added 'd' option to app_followme. This option disables the "Please hold"
1159 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
1160 received will terminate recording.
1161 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
1162 Previously the folder could only be set per context, but has now been extended
1163 using the imapfolder option.
1164 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
1165 * Voicemail now allows the pager date format to be specified separately from the
1167 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
1168 to allow joining, leaving, and sending text to group chats.
1169 * MeetMe has a new option 'G' to play an announcement before joining a conference.
1170 * Page has a new option 'A(x)' which will playback an announcement simultaneously
1171 to all paged phones (and optionally excluding the caller's one using the new
1172 option 'n') before the call is bridged.
1173 * The 'f' option to Dial has been augmented to take an optional argument. If no
1174 argument is provided, the 'f' option works as it always has. If an argument is
1175 provided, then the connected party information of all outgoing channels created
1176 during the Dial will be set to the argument passed to the 'f' option.
1177 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
1179 * The OSP lookup application adds in/outbound network ID, optional security,
1180 number portability, QoS reporting, destination IP port, custom info and service
1182 * Added new application VMSayName that will play the recorded name of the voicemail
1183 user if it exists, otherwise will play the mailbox number.
1184 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
1185 retrieve state for a particular bridge, where <name> is the conference name
1186 * app_directory now allows exiting at any time using the operator or pound key.
1187 * Voicemail now supports setting a locale per-mailbox.
1188 * Two new applications are provided for declining counting phrases in multiple
1189 languages. See the application notes for SayCountedNoun and SayCountedAdj for
1191 * Voicemail now runs the externnotify script when pollmailboxes is activated and
1193 * Voicemail now includes rdnis within msgXXXX.txt file.
1194 * ExternalIVR now supports IPv6 addresses.
1195 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
1196 at https://wiki.asterisk.org/wiki/x/oQBB
1197 * ParkedCall and Park can now specify the parking lot to use.
1201 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
1202 over SRV records associated with a specific service. From the CLI, type
1203 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
1204 details on how these may be used.
1205 * PITCH_SHIFT dialplan function added. This function can be used to modify the
1206 pitch of a channel's tx and rx audio streams.
1207 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
1208 setting various connected line and redirecting party information.
1209 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
1210 support ISDN subaddressing.
1211 * The CHANNEL() function now supports the "name" and "checkhangup" options.
1212 * For DAHDI channels, the CHANNEL() dialplan function now allows
1213 the dialplan to request changes in the configuration of the active
1214 echo canceller on the channel (if any), for the current call only.
1217 exten => s,n,Set(CHANNEL(echocan_mode)=off)
1219 The possible values are:
1221 on - normal mode (the echo canceller is actually reinitialized)
1223 fax - FAX/data mode (NLP disabled if possible, otherwise completely
1225 voice - voice mode (returns from FAX mode, reverting the changes that
1226 were made when FAX mode was requested)
1227 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
1228 and setting variables on the channel which created the current channel.
1229 Administrators should take care to avoid naming conflicts, when multiple
1230 channels are dialled at once, especially when used with the Local channel
1231 construct (which all could set variables on the master channel). Usage
1232 of the HASH() dialplan function, with the key set to the name of the slave
1233 channel, is one approach that will avoid conflicts.
1234 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
1236 * func_odbc now allows multiple row results to be retrieved without using
1237 mode=multirow. If rowlimit is set, then additional rows may be retrieved
1238 from the same query by using the name of the function which retrieved the
1239 first row as an argument to ODBC_FETCH().
1240 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
1241 dialplan. This function returns the content of the received message.
1242 * Added REPLACE, which searches a given variable name for a set of characters,
1243 then either replaces them with a single character or deletes them.
1244 * Added PASSTHRU, which literally passes the same argument back as its return
1245 value. The intent is to be able to use a literal string argument to
1246 functions that currently require a variable name as an argument.
1247 * HASH-associated variables now can be inherited across channel creation, by
1248 prefixing the name of the hash at assignment with the appropriate number of
1249 underscores, just like variables.
1250 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
1251 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
1252 whether or not channels that are bridged to the current channel will be
1253 required to have secure signaling and/or media.
1254 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
1255 the current channel has secure signaling and/or media.
1256 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
1257 "no_media_path" option.
1258 Returns "0" if there is a B channel associated with the call.
1259 Returns "1" if no B channel is associated with the call. The call is either
1260 on hold or is a call waiting call.
1261 * Added option to dialplan function CDR(), the 'f' option
1262 allows for high resolution times for billsec and duration fields.
1263 * FILE() now supports line-mode and writing.
1264 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
1265 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
1269 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
1270 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
1271 and is set when a dynamic feature is triggered.
1272 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
1273 to dynamically create a new parking lot matching the value this varible is
1275 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
1276 features.conf that should be the base for dynamic parkinglots.
1277 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
1278 parkinglot should have.
1279 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
1280 parkinglot should have.
1281 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
1286 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
1287 timeout has expired.
1288 * Added 'R' option to app_queue. This option stops moh and indicates ringing
1289 to the caller when an Agent's phone is ringing. This can be used to indicate
1290 to the caller that their call is about to be picked up, which is nice when
1291 one has been on hold for an extened period of time.
1292 * A new config option, penaltymemberslimit, has been added to queues.conf.
1293 When set this option will disregard penalty settings when a queue has too
1295 * A new option, 'I' has been added to both app_queue and app_dial.
1296 By setting this option, Asterisk will not update the caller with
1297 connected line changes or redirecting party changes when they occur.
1298 * A 'relative-periodic-announce' option has been added to queues.conf. When
1299 enabled, this option will cause periodic announce times to be calculated
1300 from the end of announcements rather than from the beginning.
1301 * The autopause option in queues.conf can be passed a new value, "all." The
1302 result is that if a member becomes auto-paused, he will be paused in all
1303 queues for which he is a member, not just the queue that failed to reach
1305 * Added dialplan function QUEUE_EXISTS to check if a queue exists
1306 * The queue logger now allows events to optionally propagate to a file,
1307 even when realtime logging is turned on. Additionally, realtime logging
1308 supports sending the event arguments to 5 individual fields, although it
1309 will fallback to the previous data definition, if the new table layout is
1312 mISDN channel driver (chan_misdn) changes
1313 ----------------------------------------
1314 * Added display_connected parameter to misdn.conf to put a display string
1315 in the CONNECT message containing the connected name and/or number if
1316 the presentation setting permits it.
1317 * Added display_setup parameter to misdn.conf to put a display string
1318 in the SETUP message containing the caller name and/or number if the
1319 presentation setting permits it.
1320 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
1321 indicate the dialplan settings are to be obtained from the asterisk
1323 * Made misdn.conf parameter callerid accept the "name" <number> format
1324 used by the rest of the system.
1325 * Made use the nationalprefix and internationalprefix misdn.conf
1326 parameters to prefix any received number from the ISDN link if that
1327 number has the corresponding Type-Of-Number. NOTE: This includes
1328 comparing the incoming call's dialed number against the MSN list.
1329 * Added the following new parameters: unknownprefix, netspecificprefix,
1330 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
1331 received number from the ISDN link if that number has the corresponding
1333 * Added new dialplan application misdn_command which permits controlling
1334 the CCBS/CCNR functionality.
1335 * Added new dialplan function mISDN_CC which permits retrieval of various
1336 values from an active call completion record.
1337 * For PTP, you should manually send the COLR of the redirected-to party
1338 for an incomming redirected call if the incoming call could experience
1339 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
1340 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
1341 if the REDIRECTING(from-num) is not empty.
1342 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
1343 option on all of the REDIRECTING statements before dialing the
1344 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
1345 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
1346 redirecting-to presentation (COLR) when it becomes available.
1347 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
1350 thirdparty mISDN enhancements
1351 -----------------------------
1352 mISDN has been modified by Digium, Inc. to greatly expand facility message
1354 * Enhanced COLP support for call diversion and transfer.
1355 * CCBS/CCNR support.
1357 The latest modified mISDN v1.1.x based version is available at:
1358 http://svn.digium.com/svn/thirdparty/mISDN/trunk
1359 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
1361 Tagged versions of the modified mISDN code are available under:
1362 http://svn.digium.com/svn/thirdparty/mISDN/tags
1363 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
1365 libpri channel driver (chan_dahdi) DAHDI changes
1366 -------------------------------------------
1367 * The channel variable PRIREDIRECTREASON is now just a status variable
1368 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
1369 to read and alter the reason.
1370 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
1371 redirected-to party for an incomming redirected call if the incoming call
1372 could experience further redirects. Just set the
1373 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
1374 to the COLR. A call has been redirected if the REDIRECTING(count) is not
1376 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
1377 use the inhibit(i) option on all of the REDIRECTING statements before
1378 dialing the redirected-to party. You still have to set the
1379 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
1380 will update the redirecting-to presentation (COLR) when it becomes available.
1381 * Added the ability to ignore calls that are not in a Multiple Subscriber
1382 Number (MSN) list for PTMP CPE interfaces.
1383 * Added dynamic range compression support for dahdi channels. It is
1384 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
1385 * Added support for ISDN calling and called subaddress with partial support
1386 for connected line subaddress.
1387 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
1388 * Added handling of received HOLD/RETRIEVE messages and the optional ability
1389 to transfer a held call on disconnect similar to an analog phone.
1390 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
1391 Will reroute/deflect an outgoing call when receive the message.
1392 Can use the DAHDISendCallreroutingFacility to send the message for the
1394 * Added standard location to add options to chan_dahdi dialing:
1395 Dial(DAHDI/g1[/extension[/options]])
1398 R Reverse charging indication
1399 * Added Reverse Charging Indication (Collect calls) send/receive option.
1400 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
1401 Dial(DAHDI/g1/extension/R)
1402 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
1403 (requires latest LibPRI)
1404 * Added ability to send/receive keypad digits in the SETUP message.
1405 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
1406 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
1407 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
1408 (requires latest LibPRI)
1409 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
1410 to eliminate tromboned calls. A tromboned call goes out an interface and comes
1411 back into the same interface. Tromboned calls happen because of call routing,
1412 call deflection, call forwarding, and call transfer.
1413 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
1414 * Added the ability to support call waiting calls. (The SETUP has no B channel
1416 * Added Malicious Call ID (MCID) event to the AMI call event class.
1417 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
1419 Asterisk Manager Interface
1420 --------------------------
1421 * The Hangup action now accepts a Cause header which may be used to
1422 set the channel's hangup cause.
1423 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
1424 to specify a separate .pem file to hold a private key. By default sslcert
1425 is used to hold both the public and private key.
1426 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
1427 for options containing the 'tls' prefix. For example, 'sslenable' is now
1428 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
1429 across all .conf files. All affected sample.conf files have been modified to
1430 reflect this change. Previous options such as 'sslenable' still work,
1431 but options with the 'tls' prefix are preferred.
1432 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
1433 in a channel. (res_mutestream.so)
1434 * The configuration file manager.conf now supports a channelvars option, which
1435 specifies a list of channel variables to include in each channel-oriented
1437 * The redirect command now has new parameters ExtraContext, ExtraExtension,
1438 and ExtraPriority to allow redirecting the second channel to a different
1439 location than the first.
1440 * Added new event "JabberStatus" in the Jabber module to monitor buddies
1442 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
1443 in a MixMonitor recording.
1444 * The 'iax2 show peers' output is now similar to the expected output of
1446 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
1448 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
1449 AOC-E messages on a channel.
1450 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
1451 conform more closely to similar events.
1452 * Added a new eventfilter option per user to allow whitelisting and blacklisting
1454 * Added optional parkinglot variable for park command.
1455 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
1456 if CallerIDNum and CallerIDName headers are also present.
1458 Channel Event Logging
1459 ---------------------
1460 * A new interface, CEL, is introduced here. CEL logs single events, much like
1461 the AMI, but it differs from the AMI in that it logs to db backends much
1462 like CDR does; is based on the event subsystem introduced by Russell, and
1463 can share in all its benefits; allows multiple backends to operate like CDR;
1464 is specialized to event data that would be of concern to billing sytems,
1465 like CDR. Backends for logging and accounting calls have been produced,
1466 but a new CDR backend is still in development.
1470 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
1471 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
1472 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
1473 * Multiple files and formats can now be specified in cdr_custom.conf.
1474 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
1475 See configs/cdr_syslog.conf.sample for more information.
1476 * A 'sequence' field has been added to CDRs which can be combined with
1477 linkedid or uniqueid to uniquely identify a CDR.
1478 * Handling of billsec and duration field has changed. If your table definition
1479 specifies those fields as float,double or similar they will now be logged with
1480 microsecond accuracy instead of a whole integer.
1482 Calendaring for Asterisk
1483 ------------------------
1484 * A new set of modules were added supporing calendar integration with Asterisk.
1485 Dialplan functions for reading from and writing to calendars are included,
1486 as well as the ability to execute dialplan logic upon calendar event notifications.
1487 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
1488 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
1489 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
1490 2003 support does not support forms-based authentication).
1492 Call Completion Supplementary Services for Asterisk
1493 ---------------------------------------------------
1494 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
1495 DAHDI/ISDN supports call completion for the following switch types:
1496 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
1497 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
1499 Multicast RTP Support
1500 ---------------------
1501 * A new RTP engine and channel driver have been added which supports Multicast RTP.
1502 The channel driver can be used with the Page application to perform multicast RTP
1503 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
1504 Type can be either basic or linksys.
1505 Destination is the IP address and port for the RTP packets.
1506 Control address is specific to the linksys type and is used for sending the control
1507 packets unique to them.
1509 Security Events Framework
1510 -------------------------
1511 * Asterisk has a new C API for reporting security events. The module res_security_log
1512 sends these events to the "security" logger level. Currently, AMI is the only
1513 Asterisk component that reports security events. However, SIP support will be
1514 coming soon. For more information on the security events framework, see the
1515 "Asterisk Security Framework" section of the Asterisk wiki at
1516 https://wiki.asterisk.org/wiki/x/wgBQ
1517 * SIP support was added in Asterisk 10
1518 * This API now supports IPv6 addresses
1522 * A technology independent fax frontend (res_fax) has been added to Asterisk.
1523 * A spandsp based fax backend (res_fax_spandsp) has been added.
1524 * The app_fax module has been deprecated in favor of the res_fax module and
1525 the new res_fax_spandsp backend.
1526 * The SendFAX and ReceiveFAX applications now send their log messages to a
1527 'fax' logger level, instead of to the generic logger levels. To see these
1528 messages, the system's logger.conf file will need to direct the 'fax' logger
1529 level to one or more destinations; the logger.conf.sample file includes an
1530 example of how to do this. Note that if the 'fax' logger level is *not*
1531 directed to at least one destination, log messages generated by these
1532 applications will be lost, and that if the 'fax' logger level is directed to
1533 the console, the 'core set verbose' and 'core set debug' CLI commands will
1534 have no effect on whether the messages appear on the console or not.
1538 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
1539 Now, in order to enable transmitting silence during record the transmit_silence
1540 option should be used. transmit_silence_during_record remains a valid option, but
1541 defaults to the behavior of the transmit_silence option.
1542 * Addition of the Unit Test Framework API for managing registration and execution
1543 of unit tests with the purpose of verifying the operation of C functions.
1544 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
1545 XMPP text messages to the remote JID.
1546 * Modules.conf has a new option - "require" - that marks a module as critical for
1547 the execution of Asterisk.
1548 If one of the required modules fail to load, Asterisk will exit with a return
1550 * An 'X' option has been added to the asterisk application which enables #exec support.
1551 This allows #exec to be used in asterisk.conf.
1552 * jabber.conf supports a new option auth_policy that toggles auto user registration.
1553 * A new lockconfdir option has been added to asterisk.conf to protect the
1554 configuration directory (/etc/asterisk by default) during reloads.
1555 * The parkeddynamic option has been added to features.conf to enable the creation
1556 of dynamic parkinglots.
1557 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
1558 the reportalarms config option.
1559 * chan_dahdi supports dialing configuring and dialing by device file name.
1560 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
1561 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
1562 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
1563 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
1564 Handy for the above name-based syntax as it does not depend on
1565 initialization order.
1566 * The Realtime dialplan switch now caches entries for 1 second. This provides a
1567 significant increase in performance (about 3X) for installations using this switchtype.
1568 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
1569 AIS. For more information, please see the Distributed Device State section of the
1570 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1571 * The addition of G.719 pass-through support.
1572 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
1573 during device configuration.
1574 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
1575 have less than 3 lines on the LCD.
1576 * Realtime now supports database failover. See the sample extconfig.conf for details.
1577 * The addition of improved translation path building for wideband codecs. Sample
1578 rate changes during translation are now avoided unless absolutely necessary.
1579 * The addition of the res_stun_monitor module for monitoring and reacting to network
1580 changes while behind a NAT.
1581 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
1582 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
1583 These allow support for any Administration. Default is AT&T values.
1587 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
1588 optionally accept a filename, to apply the setting only to the code generated from
1589 that source file when Asterisk was built. However, there are some modules in Asterisk
1590 that are composed of multiple source files, so this did not result in the behavior
1591 that users expected. In this version, 'core set debug' and 'core set verbose'
1592 can optionally accept *module* names instead (with or without the .so extension),
1593 which applies the setting to the entire module specified, regardless of which source
1594 files it was built from.
1595 * New 'manager show settings' command showing the current settings loaded from
1597 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
1598 the channel hangup request to all channels.
1599 * Added a "core reload" CLI command that executes a global reload of Asterisk.
1601 ------------------------------------------------------------------------------
1602 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
1603 ------------------------------------------------------------------------------
1607 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
1608 Snom phones use this for call pickup of extensions that the phone is
1610 * Added support for setting the domain in the URI for caller of an
1611 outbound call by using the SIPFROMDOMAIN channel variable.
1612 * Added a new configuration option "remotesecret" for authentication to
1613 remote services. For backwards compatibility, "secret" still has the
1614 same function as before, but now you can configure both a remote secret and a
1615 local secret for mutual authentication.
1616 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
1617 the sound will be played to the target of an attended transfer
1618 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
1619 finer control over how many peers Asterisk will qualify and the gap between them
1620 when all peers need to be qualified at the same time.
1621 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
1622 (either globally or for a specific peer), chan_sip will treat any SDP data
1623 it receives as new data and update the media stream accordingly. By
1624 default, Asterisk will only modify the media stream if the SDP session
1625 version received is different from the current SDP session version. This
1626 option is required to interoperate with devices that have non-standard SDP
1627 session version implementations (observed with Microsoft OCS). This option
1628 is disabled by default.
1629 * The parsing of register => lines in sip.conf has been modified to allow a port
1630 to be present in the "user" portion. Please see the sip.conf.sample file for more
1632 * Added support for subscribing to MWI on a remote server and making the status available
1633 as a mailbox. Please see the sip.conf.sample file for more information.
1634 * Added a function to remove SIP headers added in the dialplan before the
1635 first INVITE is generated - SIPRemoveHeader()
1636 * Channel variables set with setvar= in a device configuration is now
1637 set both for inbound and outbound calls.
1638 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
1642 * Added immediate option to iax.conf
1643 * Added forceencryption option to iax.conf
1644 * Added Encryption and Trunk status to manager command "iaxpeers"
1648 * The configuration file now holds separate sections for devices and lines.
1649 Please have a look at configs/skinny.conf.sample and change your skinny.conf
1654 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
1655 support for LibOpenR2. http://www.libopenr2.org/
1656 * The UK option waitfordialtone has been added for use with BT analog
1658 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
1659 is used in conjunction with the 'faxdetect' configuration option. When
1660 'faxbuffers' is used and fax tones are detected, the channel will dynamically
1661 switch to the configured faxbuffers policy. For example, to use 6 buffers
1662 and a 'full' buffer policy for a fax transmission, add:
1664 The faxbuffers configuration will be in affect until the call is torn down.
1665 * Added service message support for 4ESS/5ESS switches.
1669 * For DAHDI channels, the CHANNEL() dialplan function now
1670 supports changing the channel's buffer policy (for the current
1671 call only), using this syntax:
1673 exten => s,n,Set(CHANNEL(buffers)=6,full)
1675 This would change the channel to the 'full' buffer policy and
1676 6 (six) buffers. Possible options for this setting are the same
1677 as those in chan_dahdi.conf.
1678 * Added a new dialplan function, CURLOPT, which permits setting various
1679 options that may be useful with the CURL dialplan function, such as
1680 cookies, proxies, connection timeouts, passwords, etc.
1681 * Permit the syntax and synopsis fields of the corresponding dialplan
1682 functions to be individually set from func_odbc.conf.
1683 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
1684 * func_odbc now may specify an insert query to execute, when the write query
1685 affects 0 rows (usually indicating that no such row exists).
1686 * Added a new dialplan function, LISTFILTER, which permits removing elements
1687 from a set list, by name. Uses the same general syntax as the existing CUT
1688 and FIELDQTY dialplan functions, which also manage lists.
1689 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
1690 obtaining realtime data from the dialplan.
1691 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
1692 a subroutine when using the GoSub() and Return() applications.
1693 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
1694 of "core show function AUDIOHOOK_INHERIT" from the CLI
1695 * Added AES_ENCRYPT. For information on its use, please see the output
1696 of "core show function AES_ENCRYPT" from the CLI
1697 * Added AES_DECRYPT. For information on its use, please see the output
1698 of "core show function AES_DECRYPT" from the CLI
1699 * func_odbc now supports database transactions across multiple queries.
1703 * Scheduled meetme conferences may now have their end times extended by
1705 * app_authenticate now gives the ability to select a prompt other than
1707 * app_directory now pays attention to the searchcontexts setting in
1708 voicemail.conf and will look through all contexts, if no context is
1709 specified in the initial argument.
1710 * A new application, Originate, has been introduced, that allows asynchronous
1711 call origination from the dialplan.
1712 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
1713 in addition to the setting in the "general" context.
1714 * Added ConfBridge dialplan application which does conference bridges without
1715 DAHDI. For information on its use, please see the output of
1716 "core show application ConfBridge" from the CLI.
1720 * The Asterisk CLI has a new command, "channel redirect", which is similar in
1721 operation to the AMI Redirect action.
1722 * extensions.conf now allows you to use keyword "same" to define an extension
1723 without actually specifying an extension. It uses exactly the same pattern
1724 as previously used on the last "exten" line. For example:
1725 exten => 123,1,NoOp(something)
1726 same => n,SomethingElse()
1727 * musiconhold.conf classes of type 'files' can now use relative directory paths,
1728 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
1729 * All deprecated CLI commands are removed from the sourcecode. They are now handled
1730 by the new clialiases module. See cli_aliases.conf.sample file.
1731 * Times within timespecs are now accurate down to the minute. This is a change
1732 from historical Asterisk, which only provided timespecs rounded to the nearest
1733 even (read: evenly divisible by 2) minute mark.
1734 * The realtime switch now supports an option flag, 'p', which disables searches for
1736 * In addition to a time range and date range, timespecs now accept a 5th optional
1737 argument, timezone. This allows you to perform time checks on alternate
1738 timezones, especially if those daylight savings time ranges vary from your
1739 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
1741 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
1742 give you the correct output for an asterisk box behind nat. It will give you the
1743 externhost and localnet settings.
1744 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
1745 can connect calls in passthrough mode, as well as record and play back files.
1746 * Successful and unsuccessful call pickup can now be alerted through sounds, by
1747 using pickupsound and pickupfailsound in features.conf.
1748 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
1749 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
1750 instead of the /var/run/asterisk.pid where it used to be. This will make
1751 installs as non-root easier to manage.
1756 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
1757 be written; they will no longer be explicitly written.
1759 Asterisk Manager Interface
1760 --------------------------
1761 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
1762 a non-empty value) in your request. If you do this, any pending AMI events will
1763 *not* be included in the response to your request as they would normally, but
1764 will be left in the event queue for the next request you make to retrieve. For
1765 some applications, this will allow you to guarantee that you will only see
1766 events in responses to 'WaitEvent' actions, and can better know when to expect them.
1767 To know whether the Asterisk server supports this header or not, your client can
1768 inspect the first response back from the server to see if it includes this header:
1770 Pragma: SuppressEvents
1772 If this is included, the server supports event suppression.
1774 * Added 4 new Actions to list skinny device(s) and line(s)
1780 LDAP Schema File Additions
1781 --------------------------
1782 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
1783 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
1785 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
1786 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
1787 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
1788 * Removed redundant IPaddr (there's already IPAddress)
1789 - Gives more configuration Flags for SIP-Users available (tested)
1790 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
1791 without extensibleObject (which really should be the last resort); gives
1792 also additional possibilities for LDAP-filter
1794 ------------------------------------------------------------------------------
1795 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
1796 ------------------------------------------------------------------------------
1798 Device State Handling
1799 ---------------------
1800 * The event infrastructure in Asterisk got another big update to help support
1801 distributed events. It currently supports distributed device state and
1802 distributed Voicemail MWI (Message Waiting Indication). A new module has
1803 been merged, res_ais, which facilitates communicating events between servers.
1804 It uses the SAForum AIS (Service Availability Forum Application Interface
1805 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
1806 a cluster of Asterisk servers, and to share events between them. For more
1807 information on setting this up, refer to the Distributed Device State section
1808 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1812 * Added a new dialplan function, AST_CONFIG(), which allows you to access
1813 variables from an Asterisk configuration file.
1814 * The JACK_HOOK function now has a c() option to supply a custom client name.
1815 * Added two new dialplan functions from libspeex for audio gain control and
1816 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
1817 rx directions of a channel from the dialplan.
1818 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
1819 based on other parameters. The default is still to search based on the
1820 forwarding station ID. However, there are new options that allow you to search
1821 based on the message desk terminal ID, or the message desk number.
1822 * TIMEOUT() has been modified to be accurate down to the millisecond.
1823 * ENUM*() functions now include the following new options:
1824 - 'u' returns the full URI and does not strip off the URI-scheme.
1825 - 's' triggers ISN specific rewriting
1826 - 'i' looks for branches into an Infrastructure ENUM tree
1827 - 'd' for a direct DNS lookup without any flipping of digits.
1828 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
1829 * CHANNEL() now has options for the maximum, minimum, and standard or normal
1830 deviation of jitter, rtt, and loss for a call using chan_sip.
1832 DAHDI channel driver (chan_dahdi) Changes
1833 ----------------------------------------
1834 * Channels can now be configured using named sections in chan_dahdi.conf, just
1835 like other channel drivers, including the use of templates.
1836 * The default for pridialplan has changed from 'national' to 'unknown'.
1840 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
1841 to something that matches the pattern a hint will be created using the contents
1842 and variables evaluated.
1843 * Dialplan matching has been extended to allow an extension to return to the
1844 PBX core to wait for more digits. This is done by using the new dialplan
1845 application called "Incomplete". This will permit a whole new level of
1846 extension control, by giving the administrator more control over early
1847 matches employing one of the short-circuit pattern match operators. Note
1848 that custom applications can trigger this same behavior by returning the
1849 special value AST_PBX_INCOMPLETE.
1853 * Directory now permits both first and last names to be matched at the same
1854 time. In addition, the number of digits to enter of the name can be set in
1855 the arguments to Directory; previously, you could enter only 3, regardless
1856 of how many names are in your company. For large companies, this should be
1858 * Voicemail now permits a mailbox setting to wrap around from first to last
1859 messages, if the "messagewrap" option is set to a true value.
1860 * Voicemail now permits an external script to be run, for password validation.
1861 The script should output "VALID" or "INVALID" on stdout, depending upon the
1862 wish to validate or invalidate the password given. Arguments are:
1863 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
1865 * Dial has a new option: F(context^extension^pri), which permits a callee to
1866 continue in the dialplan, at the specified label, if the caller hangs up.
1867 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
1868 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
1869 * The Jack application now has a c() option to supply a custom client name.
1870 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
1871 like the pre-existing whisper mode, except that the spy can also talk to the
1872 participant on the bridged channel as well.
1873 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
1874 to be spoken instead of the channel name or number. For more information on the
1875 use of this option, issue the command "core show application ChanSpy" from the
1877 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
1878 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
1879 words, if using the 'd' option, it is not possible to enter a number to append to
1880 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
1881 change to whisper mode, and pressing 6 will change to barge mode.
1882 * ExternalIVR now takes several options that affect the way it performs, as
1883 well as having several new commands. Please see the External IVR page on the Asterisk
1884 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
1885 * Added ability to communicate over a TCP socket instead of forking a child process for the
1886 ExternalIVR application.
1887 * ChanIsAvail has a new option, 'a', which will return all available channels instead
1888 of just the first one if you give the function more then one channel to check.
1889 * PrivacyManager now takes an option where you can specify a context where the
1890 given number will be matched. This way you have more control over who is allowed
1891 and it stops the people who blindly enter 10 digits.
1892 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
1893 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
1894 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
1895 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
1896 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
1897 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
1898 * The Dial() application no longer copies the language used by the caller to the callee's
1899 channel. If you desire for the caller's channel's language to be used for file playback
1900 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
1901 * SendImage() no longer hangs up the channel on error; instead, it sets the
1902 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
1903 'UNSUPPORTED'. This change makes SendImage() more consistent with other
1905 * Park has a new option, 's', which silences the announcement of the parking space number.
1906 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
1907 invalid input and will be assumed to mean that no timeout is desired.
1911 * Added DNS manager support to registrations for peers referencing peer entries.
1912 DNS manager runs in the background which allows DNS lookups to be run asynchronously
1913 as well as periodically updating the IP address. These properties allow for
1914 better performance as well as recovery in the event of an IP change.
1915 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
1916 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
1917 These changes also provide performance improvements for call setup and tear down.
1918 * Added ability to specify registration expiry time on a per registration basis in
1920 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
1922 * Added t38pt_usertpsource option. See sip.conf.sample for details.
1923 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
1924 * 'sip show peers' and 'sip show users' display their entries sorted in
1925 alphabetical order, as opposed to the order they were in, in the config
1927 * Videosupport now supports an additional option, "always", which always sets
1928 up video RTP ports, even on clients that don't support it. This helps with
1929 callfiles and certain transfers to ensure that if two video phones are
1930 connected, they will always share video feeds.
1934 * Existing DNS manager lookups extended to check for SRV records.
1935 * IAX2 encryption support has been improved to support periodic key rotation
1936 within a call for enhanced security. The option "keyrotate" has been
1937 provided to disable this functionality to preserve backwards compatibility
1938 with older versions of IAX2 that do not support key rotation.
1942 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
1943 data tree based on the given <path>.
1944 * New CLI command "data show providers" that will display all the registered
1946 * New CLI command, "config reload <file.conf>" which reloads any module that
1947 references that particular configuration file. Also added "config list"
1948 which shows which configuration files are in use.
1949 * New CLI commands, "pri show version" and "ss7 show version" that will
1950 display which version of libpri and libss7 are being used, respectively.
1951 A new API call was added so trunk will now have to be compiled against
1952 a versions of libpri and libss7 that have them or it will not know that
1953 these libraries exist.
1954 * The commands "core show globals", "core set global" and "core set chanvar" has
1955 been deprecated in favor of the more semanticly correct "dialplan show globals",
1956 "dialplan set chanvar" and "dialplan set global".
1957 * New CLI command "dialplan show chanvar" to list all variables associated
1958 with a given channel.
1962 * Addresses managed by DNS manager now can check to see if there is a DNS
1963 SRV record for a given domain and will use that hostname/port if present.
1965 AMI - The manager (TCP/TLS/HTTP)
1966 --------------------------------
1967 * The Status command now takes an optional list of variables to display
1968 along with channel status.
1969 * The QueueEntry event now also includes the channel's uniqueid
1973 * res_odbc no longer has a limit of 1023 total possible unshared connections,
1974 as some people were running into this limit. This limit has been increased
1979 * The TRANSFER queue log entry now includes the the caller's original
1980 position in the transferred-from queue.
1981 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
1982 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
1983 as well as an explanation about timeout options in general
1984 * Added a new option - C - for forcing the "answered elsewhere" flag on
1985 cancellation of calls in to members of the queue. This is to avoid the
1986 call to a member of a queue having the call listed as a "missed call".
1990 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
1991 adaptive capabilities. What this means in practical terms is that if your
1992 realtime table lacks critical fields, Asterisk will now emit warnings to
1993 that effect. Also, some of the realtime drivers have the ability (if
1994 configured) to automatically add those columns to the table with the
1995 correct type and length.
1999 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
2000 the 'setvar' option to cause a given audio file to be played upon completion
2001 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
2002 Skinny channels only.
2003 * You can now compile Asterisk against the Hoard Memory Allocator, see the
2004 Hoard page on the Asterisk wiki for more information:
2005 https://wiki.asterisk.org/wiki/x/pQBB
2006 * Config file variables may now be appended to, by using the '+=' append
2007 operator. This is most helpful when working with long SQL queries in
2008 func_odbc.conf, as the queries no longer need to be specified on a single
2010 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
2011 which will add a second to the billsec when the ending
2012 time is set, if the number in the microseconds field of the end time is
2013 greater than the number of microseconds in the answer time. This allows
2014 users to count the 'initiated' seconds in their billing records.
2016 ------------------------------------------------------------------------------
2017 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
2018 ------------------------------------------------------------------------------
2020 AMI - The manager (TCP/TLS/HTTP)
2021 --------------------------------
2022 * Manager has undergone a lot of changes, all of them documented
2023 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
2024 * Manager version has changed to 1.1
2025 * Added a new action 'CoreShowChannels' to list currently defined channels
2026 and some information about them.
2027 * Added a new action 'SIPshowregistry' to list SIP registrations.
2028 * Added TLS support for the manager interface and HTTP server
2029 * Added the URI redirect option for the built-in HTTP server
2030 * The output of CallerID in Manager events is now more consistent.
2031 CallerIDNum is used for number and CallerIDName for name.
2032 * Enable https support for builtin web server.
2033 See configs/http.conf.sample for details.
2034 * Added a new action, GetConfigJSON, which can return the contents of an
2035 Asterisk configuration file in JSON format. This is intended to help
2036 improve the performance of AJAX applications using the manager interface
2038 * SIP and IAX manager events now use "ChannelType" in all cases where we
2039 indicate channel driver. Previously, we used a mixture of "Channel"
2040 and "ChannelDriver" headers.
2041 * Added a "Bridge" action which allows you to bridge any two channels that
2042 are currently active on the system.
2043 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
2044 the voicemail users setup.
2045 * Added 'DBDel' and 'DBDelTree' manager commands.
2046 * cdr_manager now reports events via the "cdr" level, separating it from
2047 the very verbose "call" level.
2048 * Manager users are now stored in memory. If you change the manager account
2049 list (delete or add accounts) you need to reload manager.
2050 * Added Masquerade manager event for when a masquerade happens between
2052 * Added "manager reload" command for the CLI
2053 * Lots of commands that only provided information are now allowed under the
2054 Reporting privilege, instead of only under Call or System.
2055 * The IAX* commands now require either System or Reporting privilege, to
2056 mirror the privileges of the SIP* commands.
2057 * Added ability to retrieve list of categories in a config file.
2058 * Added ability to retrieve the content of a particular category.
2059 * Added ability to empty a context.
2060 * Created new action to create a new file.
2061 * Updated delete action to allow deletion by line number with respect to category.
2062 * Added new action insert to add new variable to category at specified line.
2063 * Updated action newcat to allow new category to be inserted in file above another
2065 * Added new event "JitterBufStats" in the IAX2 channel
2066 * Originate now requires the Originate privilege and, if you want to call out
2067 to a subshell, it requires the System privilege, as well. This was done to
2068 enhance manager security.
2069 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
2070 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
2071 or manager show command Atxfer from the CLI
2072 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
2073 details or manager show command IAXregistry from the CLI
2077 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
2078 state in the dialplan, as well as creating custom device states that are
2079 controllable from the dialplan.
2080 * Extend CALLERID() function with "pres" and "ton" parameters to
2081 fetch string representation of calling number presentation indicator
2082 and numeric representation of type of calling number value.
2083 * MailboxExists converted to dialplan function
2084 * A new option to Dial() for telling IP phones not to count the call
2085 as "missed" when dial times out and cancels.
2086 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
2087 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
2088 held for any given channel. Also, locks are automatically freed when a
2090 * Added HINT() dialplan function that allows retrieving hint information.
2091 Hints are mappings between extensions and devices for the sake of
2092 determining the state of an extension. This function can retrieve the list
2093 of devices or the name associated with a hint.
2094 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
2096 * Added SYSINFO() dialplan function which allows retrieval of system information
2097 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
2098 the existence of a dialplan target.
2099 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
2100 upper and lower case, respectively.
2101 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
2102 ID for the call (not the Asterisk call ID or unique ID), provided that the
2103 channel driver supports this. For SIP, you get the SIP call-ID for the
2104 bridged channel which you can store in the CDR with a custom field.
2108 * Added CLI permissions, config file: cli_permissions.conf
2109 default is to allow all commands for every local user/group.
2110 Also this new feature added three new CLI commands:
2111 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
2112 - cli reload permissions
2113 - cli show permissions
2114 * New CLI command "core show hint" (usage: core show hint <exten>)
2115 * New CLI command "core show settings"
2116 * Added 'core show channels count' CLI command.
2117 * Added the ability to set the core debug and verbose values on a per-file basis.
2118 * Added 'queue pause member' and 'queue unpause member' CLI commands
2119 * Ability to set process limits ("ulimit") without restarting Asterisk
2120 * Enhanced "agi debug" to print the channel name as a prefix to the debug
2121 output to make debugging on busy systems much easier.
2122 * New CLI commands "dialplan set extenpatternmatching true/false"
2123 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
2124 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
2125 listed in the startup_commands section of cli.conf will get executed.
2126 * Added a CLI command, "devstate change", which allows you to set custom device
2127 states from the func_devstate module that provides the DEVICE_STATE() function
2128 and handling of the "Custom:" devices.
2129 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
2130 sorted into the different possible callbacks, with the number of entries
2131 currently scheduled for each. Gives you a feel for how busy the sip channel
2133 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
2134 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
2135 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
2139 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
2140 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
2141 for a received call. If it is detected, the channel will jump to the
2142 'fax' extension in the dialplan.
2143 * The default SIP useragent= identifier now includes the Asterisk version
2144 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
2145 If set, and the incoming request carries authentication info,
2146 the username to match in the users list is taken from the Digest header
2147 rather than from the From: field. This feature is considered experimental.
2148 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
2149 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
2150 * The "localmask" setting was removed in version 1.2 and the reminder about it
2151 being removed is now also removed.
2152 * A new option "busylevel" for setting a level of calls where asterisk reports
2153 a device as busy, to separate it from call-limit. This value is also added
2154 to the SIP_PEER dialplan function.
2155 * A new realtime family called "sipregs" is now supported to store SIP registration
2156 data. If this family is defined, "sippeers" will be used for configuration and
2157 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
2158 registration data, as before.
2159 * The SIPPEER function have new options for port address, call and pickup groups
2160 * Added support for T.140 realtime text in SIP/RTP
2161 * The "checkmwi" option has been removed from sip.conf, as it is no longer
2162 required due to the restructuring of how MWI is handled. See the descriptions
2163 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
2164 for more information.
2165 * Added rtpdest option to CHANNEL() dialplan function.
2166 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
2167 * SIP now adds a header to the CANCEL if the call was answered by another phone
2168 in the same dial command, or if the new c option in dial() is used.
2169 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
2170 states it is not needed. For phones, however, that do require it the "registertrying" option
2171 has been added so it can be enabled.
2172 * A new option called "callcounter" (global/peer/user level) enables call counters needed
2173 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
2174 used to enable this functionality).
2175 * New settings for timer T1 and timer B on a global level or per device. This makes it
2176 possible to force timeout faster on non-responsive SIP servers. These settings are
2177 considered advanced, so don't use them unless you have a problem.
2178 * Added a dial string option to be able to set the To: header in an INVITE to any
2180 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
2181 the qualify frequency.
2182 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
2183 were not properly torn down due to network or endpoint failures during an established
2185 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
2186 and configs/sip.conf.sample for more information on how it is used.
2187 * Added a new configuration option "authfailureevents" that enables manager events when
2188 a peer can't authenticate properly.
2189 * Added DNS manager support to registrations for peers not referencing a peer entry.
2193 * Added the trunkmaxsize configuration option to chan_iax2.
2194 * Added the srvlookup option to iax.conf
2195 * Added support for OSP. The token is set and retrieved through the CHANNEL()
2198 XMPP Google Talk/Jingle changes
2199 -------------------------------
2200 * Added the bindaddr option to gtalk.conf.
2204 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
2205 * Proper codec support in chan_skinny.
2206 * Added settings for IP and Ethernet QoS requests
2210 * Added separate settings for media QoS in mgcp.conf
2212 Console Channel Driver changes
2213 ------------------------------
2214 * Added experimental support for video send & receive to chan_oss.
2215 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
2218 Phone channel changes (chan_phone)
2219 ----------------------------------
2220 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
2222 H.323 channel Changes
2223 ---------------------
2224 * H323 remote hold notification support added (by NOTIFY message
2225 and/or H.450 supplementary service)
2227 Local channel changes
2228 ---------------------
2229 * The device state functionality in the Local channel driver has been updated
2230 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
2231 to just UNKNOWN if the extension exists.
2232 * Added jitterbuffer support for chan_local. This allows you to use the
2233 generic jitterbuffer on incoming calls going to Asterisk applications.
2234 For example, this would allow you to use a jitterbuffer for an incoming
2235 SIP call to Voicemail by putting a Local channel in the middle. This
2236 feature is enabled by using the 'j' option in the Dial string to the Local
2237 channel in conjunction with the existing 'n' option for local channels.
2238 * A 'b' option has been added which causes chan_local to return the actual channel
2239 that is behind it when queried. This is useful for transfer scenarios as the
2240 actual channel will be transferred, not the Local channel.
2242 Agent channel changes
2243 ----------------------
2244 * The ackcall and endcall options are now supplemented with options acceptdtmf
2245 and enddtmf. These allow for the DTMF keypress to be configurable. The options
2246 default to their old hard-coded values ('#' and '*' respectively) so this should
2247 not break any existing agent installations.
2249 DAHDI channel driver (chan_dahdi) Changes
2250 ----------------------------------------
2251 * SS7 support (via libss7 library)
2252 * In India, some carriers transmit CID via dtmf. Some code has been added
2253 that will handle some situations. The cidstart=polarity_IN choice has been added for
2254 those carriers that transmit CID via dtmf after a polarity change.
2255 * CID matching information is now shown when doing 'dialplan show'.
2256 * Added dahdi show version CLI command.
2257 * Added setvar support to chan_dahdi.conf channel entries.
2258 * Added two new options: mwimonitor and mwimonitornotify. These options allow
2259 you to enable MWI monitoring on FXO lines. When the MWI state changes,
2260 the script specified in the mwimonitornotify option is executed. An internal
2261 event indicating the new state of the mailbox is also generated, so that
2262 the normal MWI facilities in Asterisk work as usual.
2263 * Added signalling type 'auto', which attempts to use the same signalling type
2264 for a channel as configured in DAHDI. This is primarily designed for analog
2265 ports, but will also work for digital ports that are configured for FXS or FXO
2266 signalling types. This mode is also the default now, so if your chan_dahdi.conf
2267 does not specify signalling for a channel (which is unlikely as the sample
2268 configuration file has always recommended specifying it for every channel) then
2269 the 'auto' mode will be used for that channel if possible.
2270 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
2271 state for a channel; also ensured that the DNDState Manager event is
2272 emitted no matter how the DND state is set or cleared.
2276 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
2277 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
2278 for details. This new channel driver allows you to use Nortel i2002,
2279 i2004, and i2050 phones with Asterisk.
2280 * Added a new channel driver, chan_console, which uses portaudio as a cross
2281 platform audio interface. It was written as a channel driver that would
2282 work with Mac CoreAudio, but portaudio supports a number of other audio
2283 interfaces, as well. Note that this channel driver requires v19 or higher
2284 of portaudio; older versions have a different API.
2288 * Added the ability to specify arguments to the Dial application when using
2289 the DUNDi switch in the dialplan.
2290 * Added the ability to set weights for responses dynamically. This can be
2291 done using a global variable or a dialplan function. Using the SHELL()
2292 function would allow you to have an external script set the weight for
2294 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
2295 functions will allow you to initiate a DUNDi query from the dialplan,
2296 find out how many results there are, and access each one.
2297 * Added the ability to specifiy a port for a dundi peer.
2301 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
2302 functions will allow you to initiate an ENUM lookup from the dialplan,
2303 and Asterisk will cache the results. ENUMRESULT can be used to access
2304 the results without doing multiple DNS queries.
2308 * Added the ability to customize which sound files are used for some of the
2309 prompts within the Voicemail application by changing them in voicemail.conf
2310 * Added the ability for the "voicemail show users" CLI command to show users
2311 configured by the dynamic realtime configuration method.
2312 * MWI (Message Waiting Indication) handling has been significantly
2313 restructured internally to Asterisk. It is now totally event based
2314 instead of polling based. The voicemail application will notify other
2315 modules that have subscribed to MWI events when something in the mailbox
2317 This also means that if any other entity outside of Asterisk is changing
2318 the contents of mailboxes, then the voicemail application still needs to
2319 poll for changes. Examples of situations that would require this option
2320 are web interfaces to voicemail or an email client in the case of using
2321 IMAP storage. So, two new options have been added to voicemail.conf
2322 to account for this: "pollmailboxes" and "pollfreq". See the sample
2323 configuration file for details.
2324 * Added "tw" language support
2325 * Added support for storage of greetings using an IMAP server
2326 * Added ability to customize forward, reverse, stop, and pause keys for message playback
2327 * SMDI is now enabled in voicemail using the smdienable option.
2328 * A "lockmode" option has been added to asterisk.conf to configure the file
2329 locking method used for voicemail, and potentially other things in the
2330 future. The default is the old behavior, lockfile. However, there is a
2331 new method, "flock", that uses a different method for situations where the
2332 lockfile will not work, such as on SMB/CIFS mounts.
2333 * Added the ability to backup deleted messages, to ease recovery in the case
2334 that a user accidentally deletes a message, and discovers that they need it.
2335 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
2336 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
2337 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
2338 voicemail boxes. The SMDI interface can also poll for MWI changes when some
2339 outside entity is modifying the state of the mailbox (such as IMAP storage or
2340 a web interface of some kind).
2341 * Added the support for marking messages as "urgent." There are two methods to accomplish
2342 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
2343 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
2344 the message as urgent after he has recorded a voicemail by following the voice instructions.
2345 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
2350 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
2351 used across multiple queues.
2352 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
2353 setqueueentryvar options for each queue, see queues.conf.sample for details.
2354 * Added keepstats option to queues.conf which will keep queue
2355 statistics during a reload.
2356 * setinterfacevar option in queues.conf also now sets a variable
2357 called MEMBERNAME which contains the member's name.
2358 * Added 'Strategy' field to manager event QueueParams which represents
2359 the queue strategy in use.
2360 * Added option to run macro when a queue member is connected to a caller,
2361 see queues.conf.sample for details.
2362 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
2363 does not count paused queue members as unavailable.
2364 * Added min-announce-frequency option to queues.conf which allows you to control the
2365 minimum amount of time between queue announcements for use when the caller's queue
2366 position changes frequently.
2367 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
2369 * Added ability for non-realtime queues to have realtime members
2370 * Added the "linear" strategy to queues.
2371 * Added the "wrandom" strategy to queues.
2372 * Added new channel variable QUEUE_MIN_PENALTY
2373 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
2374 rules in queuerules.conf. See configs/queuerules.conf.sample for details
2375 * Added a new parameter for member definition, called state_interface. This may be
2376 used so that a member may be called via one interface but have a different interface's
2377 device state reported.
2378 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
2379 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
2380 "manager show command QueueReset."
2381 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
2382 specified by the periodic-announce option, then one will be chosen randomly when it is time
2383 to play a periodic announcment
2384 * New configuration options: announce-position now takes two more values in addition to "yes" and
2385 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
2386 announce-position-limit. By setting announce-position to "limit" callers will only have their
2387 position announced if their position is less than what is specified by announce-position-limit.
2388 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
2389 will be told that their are more than announce-position-limit callers waiting.
2390 * Two new queue log events have been added. An ADDMEMBER event will be logged
2391 when a realtime queue member is added and a REMOVEMEMBER event will be logged
2392 when a realtime queue member is removed. Since there is no calling channel associated
2393 with these events, the string "REALTIME" is placed where the channel's unique id
2394 is typically placed.
2395 * The configuration method for the "joinempty" and "leavewhenempty" options has
2396 changed to a comma-separated list of methods of determining member availability
2397 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
2398 values are still accepted for backwards-compatibility, though.
2399 * The average talktime is now calculated on queues. This information is reported via the
2400 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
2401 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
2406 * The 'o' option to provide an optimization has been removed and its functionality
2407 has been enabled by default.
2408 * When a conference is created, the UNIQUEID of the channel that caused it to be
2409 created is stored. Then, every channel that joins the conference will have the
2410 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
2411 callers that come and go from long standing conferences.
2412 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
2413 except it does operations on a channel by name, instead of number in a conference.
2414 This is a very useful feature in combination with the 'X' option to ChanSpy.
2415 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
2417 * Added new RealTime functionality to provide support for scheduled conferencing.
2418 This includes optional messages to the caller if they attempt to join before
2419 the schedule start time, or to allow the caller to join the conference early.
2420 Also included is optional support for limiting the number of callers per
2421 RealTime conference.
2422 * Added the S() and L() options to the MeetMe application. These are pretty
2423 much identical to the S() and L() options to Dial(). They let you set
2424 timeouts for the conference, as well as have warning sounds played to
2425 let the caller know how much time is left, and when it is running out.
2426 * Added the ability to do "meetme concise" with the "meetme" CLI command.
2427 This extends the concise capabilities of this CLI command to include
2428 listing all conferences, instead of an addition to the other sub commands
2429 for the "meetme" command.
2430 * Added the ability to specify the music on hold class used to play into the
2431 conference when there is only one member and the M option is used.
2432 * Added MEETME_INFO dialplan function which provides a way to query
2433 various properties of a Meetme conference.
2434 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
2435 and *84: record in-conf
2437 Other Dialplan Application Changes
2438 ----------------------------------
2439 * Argument support for Gosub application
2440 * From the to-do lists: straighten out the app timeout args:
2441 Wait() app now really does 0.3 seconds- was truncating arg to an int.
2442 WaitExten() same as Wait().
2443 Congestion() - Now takes floating pt. argument.
2444 Busy() - now takes floating pt. argument.
2445 Read() - timeout now can be floating pt.
2446 WaitForRing() now takes floating pt timeout arg.
2447 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
2448 * Added 's' option to Page application.
2449 * Added an optional timeout argument to the Page application.
2450 * Added 'E', 'V', and 'P' commands to ExternalIVR.
2451 * Added 'o' and 'X' options to Chanspy.
2452 * Added a new dialplan application, Bridge, which allows you to bridge the
2453 calling channel to any other active channel on the system.
2454 * Added the ability to specify a music on hold class to play instead of ringing
2455 for the SLATrunk application.
2456 * The Read application no longer exits the dialplan on error. Instead, it sets
2457 READSTATUS to ERROR, which you can catch and handle separately.
2458 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
2459 of asking for verification of each name, one at a time.
2460 * Privacy() no longer uses privacy.conf, as all options are specifyable as
2461 direct options to the app.
2462 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
2464 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
2465 * The ChannelRedirect application no longer exits the dialplan if the given channel
2466 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
2467 or NOCHANNEL if the given channel was not found.
2468 * The silencethreshold setting that was previously configurable in multiple
2469 applications is now settable globally via dsp.conf.
2471 Music On Hold Changes
2472 ---------------------
2473 * A new option, "digit", has been added for music on hold classes in
2474 musiconhold.conf. If this is set for a music on hold class, a caller
2475 listening to music on hold can press this digit to switch to listening
2476 to this music on hold class.
2477 * Support for realtime music on hold has been added.
2478 * In conjunction with the realtime music on hold, a general section has
2479 been added to musiconhold.conf, its sole variable is cachertclasses. If this
2480 is set, then music on hold classes found in realtime will be cached in memory.
2484 * AEL upgraded to use the Gosub with Arguments instead
2485 of Macro application, to hopefully reduce the problems
2486 seen with the artificially low stack ceiling that
2487 Macro bumps into. Macros can only call other Macros
2488 to a depth of 7. Tests run using gosub, show depths
2489 limited only by virtual memory. A small test demonstrated
2490 recursive call depths of 100,000 without problems.
2491 -- in addition to this, all apps that allowed a macro
2492 to be called, as in Dial, queues, etc, are now allowing
2493 a gosub call in similar fashion.
2494 * AEL now generates LOCAL(argname) declarations when it
2495 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
2496 etc. That makes the arguments local in scope. The user
2497 can define their own local variables in macros, now,
2498 by saying "local myvar=someval;" or using Set() in this
2499 fashion: Set(LOCAL(myvar)=someval); ("local" is now
2501 * utils/conf2ael introduced. Will convert an extensions.conf
2502 file into extensions.ael. Very crude and unfinished, but
2503 will be improved as time goes by. Should be useful for a
2504 first pass at conversion.
2505 * aelparse will now read extensions.conf to see if a referenced
2506 macro or context is there before issueing a warning.
2507 * AEL parser sets a local channel variable ~~EXTEN~~, to
2508 preserve the value of ${EXTEN} thru switch statements.
2509 * New operator in $[...] expressions: the ~~ operator serves
2510 as a concatenation operator. AT THE MOMENT, it is really only
2511 necessary and useful in AEL, especially in if() expressions.
2512 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
2513 any enclosing double-quotes, and evaluate to the value of a
2514 concatenated with the value of b. For example if a is set to
2515 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
2516 evaluate to xyzabc .
2519 Call Features (res_features) Changes
2520 ------------------------------------
2521 * Added the parkedcalltransfers option to features.conf
2522 * Added parkedcallparking option to control one touch parking w/ parking
2524 * Added parkedcallhangup option to control disconnect feature w/ parking
2526 * Added parkedcallrecording option to control one-touch record w/ parking
2528 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
2529 parkedcalltransfers option support for multiple parking lots.
2530 * Added BRIDGE_FEATURES variable to set available features for a channel
2531 * The built-in method for doing attended transfers has been updated to
2532 include some new options that allow you to have the transferee sent
2533 back to the person that did the transfer if the transfer is not successful.
2534 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
2535 in features.conf.sample.
2536 * Added support for configuring named groups of custom call features in
2537 features.conf. This means that features can be written a single time, and
2538 then mapped into groups of features for different key mappings or easier
2540 * Updated the ParkedCall application to allow you to not specify a parking
2541 extension. If you don't specify a parking space to pick up, it will grab
2542 the first one available.
2543 * Added cli command 'features reload' to reload call features from features.conf
2544 * Moved into core asterisk binary.
2545 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
2546 * Added the ability for custom parking lots to be configured with their own
2547 parking extension with the parkext option.
2549 Language Support Changes
2550 ------------------------
2551 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
2552 * Added support for the Hungarian language for saying numbers, dates, and times.
2556 * Added SPEECH commands for speech recognition. A complete listing can be found
2558 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
2559 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
2560 does not behave as expected; the native command needs to be used, instead.
2561 * Added the ability to perform SRV lookups on fast AGI calls. To use this
2562 feature, simply use hagi: instead of agi: as the protocol portion
2563 of the URI parameter to the AGI function call in your dial plan. Also note
2564 that specifying a port number in the AGI URI will disable SRV lookups,
2565 even if you use the hagi: protocol.
2566 * No longer support MSG_OOB flag on HANGUP.
2570 * Added rotatestrategy option to logger.conf, along with two new options:
2571 "timestamp" which will use the time to name the logger files instead of
2572 sequence number; and "rotate", which rotates the names of the log files,
2573 similar to the way syslog rotates files.
2574 * Added exec_after_rotate option to logger.conf, which allows a system
2575 command to be run after rotation. This is primarily useful with
2576 rotatestrategy=rotate, to allow a limit on the number of log files kept
2577 and to ensure that the oldest log file gets deleted.
2578 * Added realtime support for the queue log
2582 * The cdr_manager module has a [mappings] feature, like cdr_custom,
2583 to add fields to the manager event from the CDR variables.
2584 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
2585 backend database CDR table. Specifically, additional, non-standard
2586 columns are supported, merely by setting the corresponding CDR variable in
2587 your dialplan. In addition, you may alias any column to another name (for
2588 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
2589 simply "alias src => ANI" in the configuration file). Records may be
2590 posted to more than one backend, simply by specifying multiple categories
2591 in the configuration file. And finally, you may filter which CDRs get
2592 posted to each backend, by specifying a filter (which the record must
2593 match) for the particular category. Filters are additive (meaning all
2594 rules must match to post that CDR).
2595 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
2596 module. Specifically, you may add additional columns into the table and
2597 they will be set, if you set the corresponding CDR variable name. Also,
2598 if you omit columns in your database table, they will be silently skipped
2599 (but a record will still be inserted, based on what columns remain). Note
2600 that the other two features from cdr_adaptive_odbc (alias and filter) are
2601 not currently supported.
2602 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
2603 has been disabled using the NoCDR application.
2605 Miscellaneous New Modules
2606 -------------------------
2607 * Added a new CDR module, cdr_sqlite3_custom.
2608 * Added a new realtime configuration module, res_config_sqlite
2609 * Added a new codec translation module, codec_resample, which re-samples
2610 signed linear audio between 8 kHz and 16 kHz to help support wideband
2612 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
2613 based on configuration templates that use Asterisk dialplan function and
2614 variable substitution. It should be possible to create phone profiles and
2615 templates that work for the majority of phones provisioned over http. It
2616 is currently only intended to provision a single user account per phone.
2617 An example profile and set of templates for Polycom phones is provided.
2618 NOTE: Polycom firmware is not included, but should be placed in
2619 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
2620 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
2621 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
2622 provided; there is a JACK() application, and a JACK_HOOK() function. Both
2623 interfaces create an input and output JACK port. The application makes
2624 these ports the endpoint of the call. The audio coming from the channel
2625 goes out the output port and whatever comes back in on the input port is
2626 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
2627 audiohook on the channel. This lets you run the audio coming from a
2628 channel through JACK, and whatever comes back in is what gets forwarded
2629 on as the channel's audio. This is very useful for building custom
2630 vocoders or doing recording or analysis of the channel's audio in another
2632 * Added a new module, res_config_curl, which permits using a HTTP POST url
2633 to retrieve, create, update, and delete realtime information from a remote
2634 web server. Note that this module requires func_curl.so to be loaded for
2635 backend functionality.
2636 * Added a new module, res_config_ldap, which permits the use of an LDAP
2637 server for realtime data access.
2638 * Added support for writing and running your dialplan in lua using the pbx_lua
2639 module. See configs/extensions.lua.sample for examples of how to do this.
2643 * Ability to use libcap to set high ToS bits when non-root
2644 on Linux. If configure is unable to find libcap then you
2645 can use --with-cap to specify the path.
2646 * Added maxfiles option to options section of asterisk.conf which allows you to specify
2647 what Asterisk should set as the maximum number of open files when it loads.
2648 * Added the jittertargetextra configuration option.
2649 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
2650 configuration files for the IP channel drivers. The new option is "cos".
2651 This information is also documented on the Asterisk wiki at
2652 https://wiki.asterisk.org/wiki/x/EYBG
2653 * When originating a call using AMI or pbx_spool that fails the reason for failure
2654 will now be available in the failed extension using the REASON dialplan variable.
2655 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
2656 It allows you to configure a prefix for auto-monitor recordings.
2657 * A new extension pattern matching algorithm, based on a trie, is introduced
2658 here, that could noticeably speed up mid-sized to large dialplans.
2659 It is NOT used by default, as duplicating the behaviour of the old pattern
2660 matcher is still under development. A config file option, in extensions.conf,
2661 in the [general] section, called "extenpatternmatchingnew", is by default
2662 set to false; setting that to true will force the use of the new algorithm.
2663 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
2664 be used to switch the algorithms at run time.
2665 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
2666 specifying which socket to use to connect to the running Asterisk daemon
2668 * Performance enhancements to the sched facility, which is used in
2669 the channel drivers, etc. Added hashtabs and doubly-linked lists
2670 to speed up deletion; start at the beginning or end of list to
2672 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
2673 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
2674 Added regression tests to the tests/ dir, also.
2675 * Added a refcount trace feature to astobj2 for those trying to balance
2676 object creation, deletion; work, play; space and time. See the
2677 notes in astobj2.h. Also, see utils/refcounter as well, as a
2678 quick way to find unbalanced refcounts in what could be a sea
2679 of objects that were balanced.
2680 * Added logging to 'make update' command. See update.log
2681 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
2682 do not come from the remote party.
2683 * Added the 'n' option to the SpeechBackground application to tell it to not
2684 answer the channel if it has not already been answered.
2685 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
2686 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
2688 * iLBC source code no longer included (see UPGRADE.txt for details)
2689 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
2690 deadlock is detected, a backtrace of the stack which led to the lock calls
2691 will be output to the CLI.
2692 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
2693 the "core show locks" CLI command will give lock information output as well
2694 as a backtrace of the stack which led to the lock calls.
2695 * users.conf now sports an optional alternateexts property, which permits
2696 allocation of additional extensions which will reach the specified user.
2697 * A new option for the configure script, --enable-internal-poll, has been added
2698 for use with systems which may have a buggy implementation of the poll system
2699 call. If you notice odd behavior such as the CLI being unresponsive on remote
2700 consoles, you may want to try using this option. This option is enabled by default
2701 on Darwin systems since it is known that the Darwin poll() implementation has
2705 --------------------
2706 * In addition to timing from DAHDI, there is a new timing module called
2707 res_timing_timerfd. In order to use this, you must be running Linux with
2708 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
2709 script will be able to tell if you have the requirements. From menuselect, select
2710 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob