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 ------------------------------------------------------------------------------
14 AMI (Asterisk Manager Interface)
16 * The SIPqualifypeer action now acknowledges the request once it has established
17 that the request is against a known peer. It also issues a new event,
18 'SIPqualifypeerdone', once the qualify action has been completed.
20 * The PlayDTMF action now supports an optional 'Duration' parameter. This
21 specifies the duration of the digit to be played, in milliseconds.
23 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
24 updates when changes occur instead of requiring the use of pollmailboxes.
26 * CLI Command 'Manager Show Commands' no longer truncates command names longer
27 than 15 characters and no longer shows authorization requirement for commands.
28 'Manager Show Command' now displays the privileges needed for using a given
29 manager command instead.
33 * When performing queue pause/unpause on an interface without specifying an
34 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
35 least one member of any queue exists for that interface.
39 * Add queue available hint. exten => 8501,hint,Queue:markq_avail
40 Note: the suffix '_avail' after the queuename.
41 Reports 'InUse' for no logged in agents or no free agents.
42 Reports 'Idle' when an agent is free.
46 * Redirecting reasons can now be set to arbitrary strings. This means
47 that the REDIRECTING dialplan function can be used to set the redirecting
48 reason to any string. It also allows for custom strings to be read as the
49 redirecting reason from SIP Diversion headers.
51 ------------------------------------------------------------------------------
52 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
53 ------------------------------------------------------------------------------
57 * The Asterisk build system will now build and install a shared library
58 (libasteriskssl.so) used to wrap various initialization and shutdown functions
59 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
60 that Asterisk can ensure that these functions do *not* get called by any
61 modules that are loaded into Asterisk, since they should only be called once
62 in any single process. If desired, this feature can be disabled by supplying
63 the "--disable-asteriskssl" option to the configure script.
65 * A new make target, 'full', has been added to the Makefile. This performs
66 the same compilation actions as make all, but will also scan the entirety of
67 each source file for documentation. This option is needed to generate AMI
68 event documentation. Note that your system must have Python in order for
69 this make target to succeed.
71 * The optimization portion of the build system has been reworked to avoid
72 broken builds on certain architectures. All architecture-specific
73 optimization has been removed in favor of using -march=native to allow gcc
74 to detect the environment in which it is running when possible. This can
75 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
77 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
78 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
80 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
81 previously parsed the header file to obtain the version of Asterisk, you
82 will now have to go through Asterisk to get the version information.
90 * Added 'F()' option. Similar to the dial option, this can be supplied with
91 arguments indicating where the callee should go after the caller is hung up,
92 or without options specified, the priority after the Queue will be used.
97 * Added menu action admin_toggle_mute_participants. This will mute / unmute
98 all non-admin participants on a conference. The confbridge configuration
99 file also allows for the default sounds played to all conference users when
100 this occurs to be overriden using sound_participants_unmuted and
101 sound_participants_muted.
103 * Added menu action participant_count. This will playback the number of
104 current participants in a conference.
106 * Added announcement configuration option to user profile. If set the sound
107 file will be played to the user, and only the user, upon joining the
113 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
114 channels respectively before the callee channels are called.
119 * Added support for IPv6.
121 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
122 external process will cause the current playlist to be cleared, including
123 stopping any audio file that is currently playing. This is useful when you
124 want to interrupt audio playback only when specific DTMF is entered by the
130 * A new option, 'I' has been added to app_followme. By setting this option,
131 Asterisk will not update the caller with connected line changes when they
132 occur. This is similar to app_dial and app_queue.
134 * The 'N' option is now ignored if the call is already answered.
136 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
137 and caller channels respectively before the callee channels are called.
139 * The winning FollowMe outgoing call is now put on hold if the caller put it on
145 * MixMonitor hooks now have IDs associated with them which can be used to
146 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
147 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
148 now accepts that ID as an argument.
150 * Added 'm' option, which stores a copy of the recording as a voicemail in the
156 * The connect action in app_mysql now allows you to specify a port number to
157 connect to. This is useful if you run a MySQL server on a non-standard
163 * Increased the default number of allowed destinations from 5 to 12.
168 * The app_page application now no longer depends on DAHDI or app_meetme. It
169 has been re-architected to use app_confbridge internally.
174 * Added queue options autopausebusy and autopauseunavail for automatically
175 pausing a queue member when their device reports busy or congestion.
177 * The 'ignorebusy' option for queue members has been deprecated in favor of
178 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
179 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
180 per interface basis. Individual ringinuse values can now be set in
181 queues.conf via an argument to member definitions. Lastly, the queue
182 'ringinuse' setting now only determines defaults for the per member
183 'ringinuse' setting and does not override per member settings like it does
186 * Added 'F()' option. Similar to the dial option, this can be supplied with
187 arguments indicating where the callee should go after the caller is hung up,
188 or without options specified, the priority after the Queue will be used.
190 * Added new option log_member_name_as_agent, which will cause the membername to
191 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
192 state_interface has been set.
194 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
198 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
199 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
200 changed arguments to SayUnixTime so that every option is truly optional even
201 when using multiple options (so that j option could be used without having to
202 manually specify timezone and format) There are other benefits, e.g., format
203 can now be used without specifying time zone as well.
208 * Addition of the VM_INFO function - see Function changes.
210 * The imapserver, imapport, and imapflags configuration options can now be
211 overriden on a user by user basis.
213 * When voicemail plays a message's envelope with saycid set to yes, when
214 reaching the caller id field it will play a recording of a file with the same
215 base name as the sender's callerid if there is a similarly named file in
216 <astspooldir>/recordings/callerids/
218 * Voicemails now contains a unique message identifier "msg_id", which is stored
219 in the message envelope with the sound files. IMAP backends will now store
220 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
221 backends will store the message identifier in a "msg_id" column. See
222 UPGRADE.txt for more information.
224 * Added VoiceMailPlayMsg application. This application will play a single
225 voicemail message from a mailbox. The result of the application, SUCCESS or
226 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
231 * Hangup handlers can be attached to channels using the CHANNEL() function.
232 Hangup handlers will run when the channel is hung up similar to the h
233 extension. The hangup_handler_push option will push a GoSub compatible
234 location in the dialplan onto the channel's hangup handler stack. The
235 hangup_handler_pop option will remove the last added location, and optionally
236 replace it with a new GoSub compatible location. The hangup_handler_wipe
237 option will remove all locations on the stack, and optionally add a new
240 * The expression parser now recognizes the ABS() absolute value function,
241 which will convert negative floating point values to positive values.
243 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
244 control of faxdetect.
246 * Addition of the VM_INFO function that can be used to retrieve voicemail
247 user information, such as the email address and full name.
248 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
251 * The REDIRECTING function now supports the redirecting original party id
254 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
255 lets you set some of the configuration options from the [general] section
256 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
257 the key sequence used to activate built-in features, such as blindxfer,
258 and automon. See the built-in documentation for details.
260 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
261 instead of simply the uri. This is the format that MessageSend() can use
262 in the from parameter for outgoing SIP messages.
264 * Added the PRESENCE_STATE function. This allows retrieving presence state
265 information from any presence state provider. It also allows setting
266 presence state information from a CustomPresence presence state provider.
267 See AMI/CLI changes for related commands.
269 * Added the AMI_CLIENT function to make manager account attributes available
270 to the dialplan. It currently supports returning the current number of
271 active sessions for a given account.
273 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
274 and the REDIRECTING functions.
282 * Added a manager event "LocalBridge" for local channel call bridges between
283 the two pseudo-channels created.
288 * Added dialtone_detect option for analog ports to disconnect incoming
289 calls when dialtone is detected.
291 * Added option colp_send to send ISDN connected line information. Allowed
292 settings are block, to not send any connected line information; connect, to
293 send connected line information on initial connect; and update, to send
294 information on any update during a call. Default is update.
296 * Add options namedcallgroup and namedpickupgroup to support installations
297 where a higher number of groups (>64) is required.
299 * Added support to use private party ID information with PRI calls.
304 * A new channel driver named chan_motif has been added which provides support for
305 Google Talk and Jingle in a single channel driver. This new channel driver includes
306 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
307 hold, unhold, and ringing notification. It is also compliant with the current Jingle
308 specification, current Google Jingle specification, and the original Google Talk
314 * Added NAT support for RTP. Setting in config is 'nat', which can be set
315 globally and overriden on a peer by peer basis.
317 * Direct media functionality has been added. Options in config are:
318 directmedia (directrtp) and directrtpsetup (earlydirect)
320 * ChannelUpdate events now contain a CallRef header.
325 * Asterisk will no longer substitute CID number for CID name in the display
326 name field if CID number exists without a CID name. This change improves
327 compatibility with certain device features such as Avaya IP500's directory
330 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
331 created using that setting to not be removed during SIP reload.
333 * Added settings recordonfeature and recordofffeature. When receiving an INFO
334 request with a "Record:" header, this will turn the requested feature on/off.
335 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
336 dynamic features must be enabled and configured properly on the requesting
337 channel for this to function properly.
339 * Add support to realtime for the 'callbackextension' option.
341 * When multiple peers exist with the same address, but differing
342 callbackextension options, incoming requests that are matched by address
343 will be matched to the peer with the matching callbackextension if it is
346 * Two new NAT options, auto_force_rport and auto_comedia, have been added
347 which set the force_rport and comedia options automatically if Asterisk
348 detects that an incoming SIP request crossed a NAT after being sent by
351 * NAT settings are now a combinable list of options. The equivalent of the
352 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
354 * Adds an option send_diversion which can be disabled to prevent
355 diversion headers from automatically being added to INVITE requests.
357 * Add support for lightweight NAT keepalive. If enabled a blank packet will
358 be sent to the remote host at a given interval to keep the NAT mapping open.
359 This can be enabled using the keepalive configuration option.
361 * Add option 'tonezone' to specify country code for indications. This option
362 can be set both globally and overridden for specific peers.
364 * The SIP Security Events Framework now supports IPv6.
366 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
367 between multiple user agents. When set, for directmedia reinvites,
368 Asterisk will not send an immediate reinvite on an incoming call leg. This
369 option is useful when peered with another SIP user agent that is known to
370 send immediate direct media reinvites upon call establishment.
372 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
375 * Add options subminexpiry and submaxexpiry to set limits of subscription
376 timer independently from registration timer settings. The setting of the
377 registration timer limits still is done by options minexpiry, maxexpiry
378 and defaultexpiry. For backwards compatibility the setting of minexpiry
379 and maxexpiry also is used to configure the subscription timer limits if
380 subminexpiry and submaxexpiry are not set in sip.conf.
382 * Set registration timer limits to default values when reloading sip
383 configuration and values are not set by configuration.
385 * Add options namedcallgroup and namedpickupgroup to support installations
386 where a higher number of groups (>64) is required.
388 * When a MESSAGE request is received, the address the request was received from
389 is now saved in the SIP_RECVADDR variable.
391 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
392 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
393 the ANI2/OLI information is set on the channel, which can be retrieved using
394 the CALLERID function.
396 * Peers can now be configured to support negotiation of ICE candidates using
397 the setting icesupport. See res_rtp_asterisk changes for more information.
399 * Added support for format attribute negotiation. See the Codecs changes for
402 * Extra headers specified with SIPAddHeader are sent with the REFER message
403 when using Transfer application. See refer_addheaders in sip.conf.sample.
405 * Added support to use private party ID information with calls.
410 * Added skinny version 17 protocol support.
415 * Added ability to use multiple lines for a single phone. This allows multiple
416 calls to occur on a single phone, using callwaiting and switching between calls.
418 * Added option 'sharpdial' allowing end dialing by pressing # key
420 * Added option 'interdigit_timer' to control phone dial timeout
422 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
424 * Added global 'debug' option, that enables debug in channel driver
426 * Added ability to translate on-screen menu in multiple languages. Tested on
427 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
428 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
431 * In addition to English added French and Russian languages for on-screen menus
433 * Reworked dialing number input: added dialing by timeout, immediate dial on
434 on dialplan compare, phone number length now not limited by screen size
436 * Added ability to pickup a call using features.conf defined value and
442 * Add options namedcallgroup and namedpickupgroup to support installations
443 where a higher number of groups (>64) is required.
445 * Added support to use private party ID information with calls.
450 * The minimum DTMF duration can now be configured in asterisk.conf
451 as "mindtmfduration". The default value is (as before) set to 80 ms.
452 (previously it was only available in source code)
454 * Named ACLs can now be specified in acl.conf and used in configurations that
455 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
456 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
457 working ACL. In addition, some CLI commands have been added to provide
458 show information and allow for module reloading - see CLI Changes.
460 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
461 items (separated by commas), and items in the rule can be negated by prefixing
462 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
463 longer necessray to control the order that the 'permit' and 'deny' columns are
464 returned from queries.
466 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
467 be used within the dynamic weight attribute when specifying a mapping.
469 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
470 header, instead of putting the user defined event name there. When enabled
471 the UserDefType header is added for user defined events. This feature is
472 enabled with the setting show_user_defined.
474 * Macro has been deprecated in favor of GoSub. For redirecting and connected
475 line purposes use the following variables instead of their macro equivalents:
476 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
477 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
478 cc_callback_macro in channel configurations.
480 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
483 * Call files now support the "early_media" option to connect with an outgoing
484 extension when early media is received.
486 * Added support to use private party ID information with calls.
491 * A new channel variable, AGIEXITONHANGUP, has been added which allows
492 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
493 AGI application would exit immediately after a channel hangup is detected.
495 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
496 are resolved and each address is attempted in turn until one succeeds or
500 AMI (Asterisk Manager Interface)
502 * The originate action now has an option "EarlyMedia" that enables the
503 call to bridge when we get early media in the call. Previously,
504 early media was disregarded always when originating calls using AMI.
506 * Added setvar= option to manager accounts (much like sip.conf)
508 * Originate now generates an error response if the extension given is not found
511 * MixMonitor will now show IDs associated with the mixmonitor upon creating
512 them if the i(variable) option is used. StopMixMonitor will accept
513 MixMonitorID as an option to close specific MixMonitors.
515 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
516 updated to include information about peers configured with
517 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
518 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
519 returned if auto_force_rport is not enabled.
521 * Added SIPpeerstatus manager command which will generate PeerStatus events
522 similar to the existing PeerStatus events found in chan_sip on demand.
524 * Hangup now can take a regular expression as the Channel option. If you want
525 to hangup multiple channels, use /regex/ as the Channel option. Existing
526 behavior to hanging up a single channel is unchanged, but if you pass a regex,
527 the manager will send you a list of channels back that were hung up.
529 * Support for IPv6 addresses has been added.
531 * AMI Events can now be documented in the Asterisk source. Note that AMI event
532 documentation is only generated when Asterisk is compiled using 'make full'.
533 See the CLI section for commands to display AMI event information.
535 * The AMI Hangup event now includes the AccountCode header so you can easily
536 correlate with AMI Newchannel events.
538 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
539 the StateInterface of the queue member.
541 * Added AMI event SessionTimeout in the Call category that is issued when a
542 call is terminated due to either RTP stream inactivity or SIP session timer
545 * CEL events can now contain a user defined header UserDefType. See core
546 changes for more information.
548 * OOH323 ChannelUpdate events now contain a CallRef header.
550 * Added PresenceState command. This command will report the presence state for
551 the given presence provider.
553 * Added Parkinglots command. This will list all parking lots as a series of
554 AMI Parkinglot events.
556 * Added MessageSend command. This behaves in the same manner as the
557 MessageSend application, and is a technolgoy agnostic mechanism to send out
558 of call text messages.
560 * Added "message" class authorization. This grants an account permission to
561 send out of call messages. Write-only.
566 * The "dialplan add include" command has been modified to create context a context
567 if one does not already exist. For instance, "dialplan add include foo into bar"
568 will create context "bar" if it does not already exist.
570 * A "dialplan remove context" command has been added to remove a context from
573 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
574 filenames of all running mixmonitors on a channel.
576 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
577 numeric instead of 0, 1, or 2.
579 * "stun show status" will show a table describing how the STUN client is
582 * "acl show [named acl]" will show information regarding a Named ACL. The
583 acl module can be reloaded with "reload acl".
585 * Added CLI command to display AMI event information - "manager show events",
586 which shows a list of all known and documented AMI events, and "manager show
587 event [event name]", which shows detail information about a specific AMI
590 * The result of the CLI command "queue show" now includes the state interface
591 information of the queue member.
593 * The command "core set verbose" will now set a separate level of logging for
594 each remote console without affecting any other console.
596 * Added command "cdr show pgsql status" to check connection status
598 * "sip show channel" will now display the complete route set.
600 * Added "presencestate list" command. This command will list all custom
601 presence states that have been set by using the PRESENCE_STATE dialplan
604 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
605 command. This changes a custom presence to a new state.
610 * Codec lists may now be modified by the '!' character, to allow succinct
611 specification of a list of codecs allowed and disallowed, without the
612 requirement to use two different keywords. For example, to specify all
613 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
615 * Add support for parsing SDP attributes, generating SDP attributes, and
616 passing it through. This support includes codecs such as H.263, H.264, SILK,
617 and CELT. You are able to set up a call and have attribute information pass.
618 This should help considerably with video calls.
620 * The iLBC codec can now use a system-provided iLBC library if one is installed,
621 just like the GSM codec.
625 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
626 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
630 * Asterisk version and build information is now logged at the beginning of a
633 * Threads belonging to a particular call are now linked with callids which get
634 added to any log messages produced by those threads. Log messages can now be
635 easily identified as involved with a certain call by looking at their call id.
636 Call ids may also be attached to log messages for just about any case where
637 it can be determined to be related to a particular call.
639 * Each logging destination and console now have an independent notion of the
640 current verbosity level. Logger.conf now allows an optional argument to
641 the 'verbose' specifier, indicating the level of verbosity sent to that
642 particular logging destination. Additionally, remote consoles now each
643 have their own verbosity level. The command 'core set verbose' will now set
644 a separate level for each remote console without affecting any other
650 * Added 'announcement' option which will play at the start of MOH and between
651 songs in modes of MOH that can detect transitions between songs (eg.
657 * New per parking lot options: comebackcontext and comebackdialtime. See
658 configs/features.conf.sample for more details.
660 * Channel variable PARKER is now set when comebacktoorigin is disabled in
663 * Channel variable PARKEDCALL is now set with the name of the parking lot
664 when a timeout occurs.
670 CDR Postgresql Driver
672 * Added command "cdr show pgsql status" to check connection status
675 CDR Adaptive ODBC Driver
677 * Added schema option for databases that support specifying a schema.
685 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
686 CALENDAR_WRITE has completed successfully.
691 * A new option, 'probation' has been added to rtp.conf
692 RTP in strictrtp mode can now require more than 1 packet to exit learning
693 mode with a new source (and by default requires 4). The probation option
694 allows the user to change the required number of packets in sequence to any
695 desired value. Use a value of 1 to essentially restore the old behavior.
696 Also, with strictrtp on, Asterisk will now drop all packets until learning
697 mode has successfully exited. These changes are based on how pjmedia handles
698 media sources and source changes.
700 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
701 enabled or disabled using the icesupport setting. A variety of other
702 settings have been introduced to configure STUN/TURN connections.
707 * A new module, res_corosync, has been introduced. This module uses the
708 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
709 of Asterisk servers to both Message Waiting Indication (MWI) and/or
710 Device State (presence) information. This module is very similar to, and
711 is a replacement for the res_ais module that was in previous releases of
717 * This module adds a cleaned up, drop-in replacement for res_jabber called
718 res_xmpp. This provides the same externally facing functionality but is
719 implemented differently internally. res_jabber has been deprecated in favor
720 of res_xmpp; please see the UPGRADE.txt file for more information.
725 * The safe_asterisk script has been updated to allow several of its parameters
726 to be set from environment variables. This also enables a custom run
727 directory of Asterisk to be specified, instead of defaulting to /tmp.
729 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
730 its value to determine the directory to assume is the top-level directory of
731 the source tree. If the variable is not set, it defaults to the current
732 behavior and uses the current working directory.
734 ------------------------------------------------------------------------------
735 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
736 ------------------------------------------------------------------------------
740 * Asterisk now has protocol independent support for processing text messages
741 outside of a call. Messages are routed through the Asterisk dialplan.
742 SIP MESSAGE and XMPP are currently supported. There are options in
743 jabber.conf and sip.conf to allow enabling these features.
744 -> jabber.conf: see the "sendtodialplan" and "context" options.
745 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
746 and "outofcall_message_context" options.
747 The MESSAGE() dialplan function and MessageSend() application have been
748 added to go along with this functionality. More detailed usage information
749 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
750 * If real-time text support (T.140) is negotiated, it will be preferred for
751 sending text via the SendText application. For example, via SIP, messages
752 that were once sent via the SIP MESSAGE request would be sent via RTP if
753 T.140 text is negotiated for a call.
757 * parkedmusicclass can now be set for non-default parking lots.
759 Asterisk Manager Interface
760 --------------------------
761 * PeerStatus now includes Address and Port.
762 * Added Hold events for when the remote party puts the call on and off hold
763 for chan_dahdi ISDN channels.
764 * Added new action MeetmeListRooms to list active conferences (shows same
765 data as "meetme list" at the CLI).
766 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
767 Description field that is set by 'description' in the channel configuration
769 * Added Uniqueid header to UserEvent.
770 * Added new action FilterAdd to control event filters for the current session.
771 This requires the system permission and uses the same filter syntax as
772 filters that can be defined in manager.conf
773 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
774 versions had some instances of the event converted, but others were left
775 as-is. All Unlink events should now be converted to Bridge events. The AMI
776 protocol version number was incremented to 1.2 as a result of this change.
779 --------------------------
780 * The HTTP Server can bind to IPv6 addresses.
783 --------------------------
784 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
785 with busydetect. usage example: busypattern=200,200,200,600
788 --------------------------
789 * New 'gtalk show settings' command showing the current settings loaded from
791 * The 'logger reload' command now supports an optional argument, specifying an
792 alternate configuration file to use.
793 * 'dialplan add extension' command will now automatically create a context if
794 the specified context does not exist with a message indicated it did so.
795 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
796 Description field which can be populated with 'description' in the channel
797 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
800 --------------------------
801 * The filter option in cdr_adaptive_odbc now supports negating the argument,
802 thus allowing records which do NOT match the specified filter.
803 * Added ability to log CONGESTION calls to CDR
806 --------------------------
807 * Ability to define custom SILK formats in codecs.conf.
808 * Addition of speex32 audio format with translation.
809 * CELT codec pass-through support and ability to define
810 custom CELT formats in codecs.conf.
811 * Ability to read raw signed linear files with sample rates
812 ranging from 8khz - 192khz. The new file extensions introduced
813 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
814 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
815 Skinny, H.323, etc) can still only support the following codecs:
816 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
817 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
818 Video: h261, h263, h263p, h264, mpeg4
823 --------------------------
824 * New highly optimized and customizable ConfBridge application capable of
825 mixing audio at sample rates ranging from 8khz-96khz.
826 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
827 and bridge profiles on a channel.
828 * CONFBRIDGE_INFO dialplan function capable of retrieving information
829 about a conference such as locked status and number of parties, admins,
831 * Addition of video_mode option in confbridge.conf for adding video support
832 into a bridge profile.
833 * Addition of the follow_talker video_mode in confbridge.conf. This video
834 mode dynamically switches the video feed to always display the loudest talker
835 supplying video in the conference.
839 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
840 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
841 variables from asterisk.conf.
845 * Addition of the JITTERBUFFER dialplan function. This function allows
846 for jitterbuffering to occur on the read side of a channel. By using
847 this function conference applications such as ConfBridge and MeetMe can
848 have the rx streams jitterbuffered before conference mixing occurs.
849 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
851 * Added STRREPLACE function. This function let's the user search a variable
852 for a given string to replace with another string as many times as the
853 user specifies or just throughout the whole string.
854 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
855 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
856 * Added extensions to chan_ooh323 in function CHANNEL()
858 libpri channel driver (chan_dahdi) DAHDI changes
859 --------------------------
860 * Added moh_signaling option to specify what to do when the channel's bridged
861 peer puts the ISDN channel on hold.
862 * Added display_send and display_receive options to control how the display ie
863 is handled. To send display text from the dialplan use the SendText()
864 application when the option is enabled.
865 * Added mcid_send option to allow sending a MCID request on a span.
868 --------------------------
869 * Added setvar option to calendar.conf to allow setting channel variables on
870 notification channels.
871 * Added "calendar show types" CLI command to list registered calendar
875 --------------------------
876 * Added two new options, r and t with file name arguments to record
877 single direction (unmixed) audio recording separate from the bidirectional
878 (mixed) recording. The mixed file name argument is optional now as long
879 as at least one recording option is used.
882 --------------------------
883 * Added a new option, l, which will disable local call optimization for
884 channels involved with the FollowMe thread. Use this option to improve
885 compatability for a FollowMe call with certain dialplan apps, options, and
889 --------------------------
890 * Added option "k" that will automatically close the conference when there's
891 only one person left when a user exits the conference.
894 --------------------------
895 * cel_pgsql now supports the 'extra' column for data added using the
896 CELGenUserEvent() application.
899 --------------------------
900 * Support for defining hints has been added to pbx_lua. See the 'hints' table
901 in the sample extensions.lua file for syntax details.
902 * Applications that perform jumps in the dialplan such as Goto will now
903 execute properly. When pbx_lua detects that the context, extension, or
904 priority we are executing on has changed it will immediately return control
905 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
906 the priority after the currently executing priority.
907 * An autoservice is now started by default for pbx_lua channels. It can be
908 stopped and restarted using the autoservice_stop() and autoservice_start()
912 --------------------------
913 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
914 into a FAXStatus event with an 'Operation' header that will be either
915 'send', 'receive', and 'gateway'.
916 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
917 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
918 feature will handle converting a fax call between an audio T.30 fax terminal
919 and an IFP T.38 fax terminal.
923 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
924 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
925 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
929 * Added general option negative_penalty_invalid default off. when set
930 members are seen as invalid/logged out when there penalty is negative.
931 for realtime members when set remove from queue will set penalty to -1.
932 * Added queue option autopausedelay when autopause is enabled it will be
933 delayed for this number of seconds since last successful call if there
934 was no prior call the agent will be autopaused immediately.
935 * Added member option ignorebusy this when set and ringinuse is not
936 will allow per member control of multiple calls as ringinuse does for
938 * Added global option check_state_unknown to enforce checking of device state
939 when the device state is unknown app_queue will see unknown as available.
943 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
945 * Added 'k' option to MeetMe to automatically kill the conference when there's only
946 one participant left (much like a normal call bridge)
947 * Added extra argument to Originate to set timeout.
951 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
952 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
953 utility in the UTILS section of menuselect. If an existing astdb is found and no
954 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
955 convert an existing astdb to the SQLite3 version automatically at runtime.
959 * Modules marked as deprecated are no longer marked as building by default. Enabling
960 these modules is still available via menuselect.
964 * authdebug is now disabled by default. To enable this functionaility again
965 set authdebug = yes in iax.conf.
969 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
970 releases it was disabled.
974 * The PBX core previously made a call with a non-existing extension test for
975 extension s@default and jump there if the extension existed.
976 This was a bad default behaviour and violated the principle of least surprise.
977 It has therefore been changed in this release. It may affect some
978 applications and configurations that rely on this behaviour. Most channel
979 drivers have avoided this for many releases by testing whether the extension
980 called exists before starting the PBX and generating a local error.
981 This behaviour still exists and works as before.
983 Extension "s" is used when no extension is given in a channel driver,
984 like immediate answer in DAHDI or calling to a domain with no user part
987 ------------------------------------------------------------------------------
988 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
989 ------------------------------------------------------------------------------
993 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
994 now defaults to force_rport. It is very important that phones requiring nat=no be
995 specifically set as such instead of relying on the default setting. If at all
996 possible, all devices should have nat settings configured in the general section as
997 opposed to configuring nat per-device.
998 * Added preferred_codec_only option in sip.conf. This feature limits the joint
999 codecs sent in response to an INVITE to the single most preferred codec.
1000 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
1001 to be used for the outgoing call. It must be one of the codecs configured
1003 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
1004 to be used for holding a private key. If tlsprivatekey is not specified,
1005 tlscertfile is searched for both public and private key.
1006 * Added tlsclientmethod option to sip.conf. This allows the protocol for
1007 outbound client connections to be specified.
1008 * The sendrpid parameter has been expanded to include the options
1009 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
1010 header to be sent (equivalent to setting sendrpid=yes) and setting
1011 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
1012 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
1013 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
1014 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
1015 will accept the SDP even if the SDP version number is not properly incremented,
1016 but will generate a warning in the log indicating that the SIP peer that sent
1017 the SDP should have the 'ignoresdpversion' option set.
1018 * The 'nat' option has now been been changed to have yes, no, force_rport, and
1019 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
1020 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
1021 remote side requests it and disables symmetric RTP support. Setting it to
1022 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
1023 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
1024 and enables symmetric RTP support.
1025 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
1026 response. This permits the master channel to know how each channel dialled
1027 in a multi-channel setup resolved in an individual way. This carries a
1028 performance penalty and can be disabled in sip.conf using the
1029 'storesipcause' option.
1030 * Added 'externtcpport' and 'externtlsport' options to allow custom port
1031 configuration for the externip and externhost options when tcp or tls is used.
1032 * Added support for message body (stored in content variable) to SIP NOTIFY message
1033 accessible via AMI and CLI.
1034 * Added 'media_address' configuration option which can be used to explicitly specify
1035 the IP address to use in the SDP for media (audio, video, and text) streams.
1036 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
1037 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
1039 * Added 'use_q850_reason' configuration option for generating and parsing
1040 if available Reason: Q.850;cause=<cause code> header. It is implemented
1041 in some gateways for better passing PRI/SS7 cause codes via SIP.
1042 * When dialing SIP peers, a new component may be added to the end of the dialstring
1043 to indicate that a specific remote IP address or host should be used when dialing
1044 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
1045 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
1046 ability to selectively force bridged channels to also be encrypted is also
1047 implemented. Branching in the dialplan can be done based on whether or not
1048 a channel has secure media and/or signaling.
1049 * Added directmediapermit/directmediadeny to limit which peers can send direct media
1051 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
1052 Charge messages to snom phones.
1053 * Added support for G.719 media streams.
1054 * Added support for 16khz signed linear media streams.
1055 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
1056 RTP has been outfitted with the same abilities.
1057 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
1058 available in device configurations as well as in the dial plan.
1059 * Addition of the 'subscribe_network_change' option for turning on and off
1060 res_stun_monitor module support in chan_sip.
1061 * Addition of the 'auth_options_requests' option for turning on and off
1062 authentication for OPTIONS requests in chan_sip.
1066 * Add #tryinclude statement for config files. This provides the same
1067 functionality as the #include statement however an asterisk module will
1068 still load if the filename does not exist. Using the #include statement
1069 Asterisk will not allow the module to load.
1073 * Added rtsavesysname option into iax.conf to allow the systname to be saved
1074 on realtime updates.
1075 * Added the ability for chan_iax2 to inform the dialplan whether or not
1076 encryption is being used. This interoperates with the SIP SRTP implementation
1077 so that a secure SIP call can be bridged to a secure IAX call when the
1078 dialplan requires bridged channels to be "secure".
1079 * Addition of the 'subscribe_network_change' option for turning on and off
1080 res_stun_monitor module support in chan_iax.
1085 * Added ability to preset channel variables on indicated lines with the setvar
1086 configuration option. Also, clearvars=all resets the list of variables back
1088 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
1089 See configs/res_pktccops.conf for more information.
1091 XMPP Google Talk/Jingle changes
1092 -------------------------------
1093 * Added the externip option to gtalk.conf.
1094 * Added the stunaddr option to gtalk.conf which allows for the automatic
1095 retrieval of the external ip from a stun server.
1099 * Added 'p' option to PickupChan() to allow for picking up channel by the first
1100 match to a partial channel name.
1101 * Added .m3u support for Mp3Player application.
1102 * Added progress option to the app_dial D() option. When progress DTMF is
1103 present, those values are sent immediately upon receiving a PROGRESS message
1104 regardless if the call has been answered or not.
1105 * Added functionality to the app_dial F() option to continue with execution
1106 at the current location when no parameters are provided.
1107 * Added the 'a' option to app_dial to answer the calling channel before any
1108 announcements or macros are executed.
1109 * Modified app_dial to set answertime when the called channel answers even if
1110 the called channel hangs up during playback of an announcement.
1111 * Modified app_dial 'r' option to support an additional parameter to play an
1112 indication tone from indications.conf
1113 * Added c() option to app_chanspy. This option allows custom DTMF to be set
1114 to cycle through the next available channel. By default this is still '*'.
1115 * Added x() option to app_chanspy. This option allows DTMF to be set to
1116 exit the application.
1117 * The Voicemail application has been improved to automatically ignore messages
1118 that only contain silence.
1119 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
1120 associated mailbox(es) to be greetings-only.
1121 * The ChanSpy application now has the 'S' option, which makes the application
1122 automatically exit once it hits a point where no more channels are available
1124 * The ChanSpy application also now has the 'E' option, which spies on a single
1125 channel and exits when that channel hangs up.
1126 * The MeetMe application now turns on the DENOISE() function by default, for
1127 each participant. In our tests, this has significantly decreased background
1128 noise (especially noisy data centers).
1129 * Voicemail now permits storage of secrets in a separate file, located in the
1130 spool directory of each individual user. The control for this is located in
1131 the "passwordlocation" option in voicemail.conf. Please see the sample
1132 configuration for more information.
1133 * The ChanIsAvail application now exposes the returned cause code using a separate
1134 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
1135 * Added 'd' option to app_followme. This option disables the "Please hold"
1137 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
1138 received will terminate recording.
1139 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
1140 Previously the folder could only be set per context, but has now been extended
1141 using the imapfolder option.
1142 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
1143 * Voicemail now allows the pager date format to be specified separately from the
1145 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
1146 to allow joining, leaving, and sending text to group chats.
1147 * MeetMe has a new option 'G' to play an announcement before joining a conference.
1148 * Page has a new option 'A(x)' which will playback an announcement simultaneously
1149 to all paged phones (and optionally excluding the caller's one using the new
1150 option 'n') before the call is bridged.
1151 * The 'f' option to Dial has been augmented to take an optional argument. If no
1152 argument is provided, the 'f' option works as it always has. If an argument is
1153 provided, then the connected party information of all outgoing channels created
1154 during the Dial will be set to the argument passed to the 'f' option.
1155 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
1157 * The OSP lookup application adds in/outbound network ID, optional security,
1158 number portability, QoS reporting, destination IP port, custom info and service
1160 * Added new application VMSayName that will play the recorded name of the voicemail
1161 user if it exists, otherwise will play the mailbox number.
1162 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
1163 retrieve state for a particular bridge, where <name> is the conference name
1164 * app_directory now allows exiting at any time using the operator or pound key.
1165 * Voicemail now supports setting a locale per-mailbox.
1166 * Two new applications are provided for declining counting phrases in multiple
1167 languages. See the application notes for SayCountedNoun and SayCountedAdj for
1169 * Voicemail now runs the externnotify script when pollmailboxes is activated and
1171 * Voicemail now includes rdnis within msgXXXX.txt file.
1172 * ExternalIVR now supports IPv6 addresses.
1173 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
1174 at https://wiki.asterisk.org/wiki/x/oQBB
1175 * ParkedCall and Park can now specify the parking lot to use.
1179 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
1180 over SRV records associated with a specific service. From the CLI, type
1181 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
1182 details on how these may be used.
1183 * PITCH_SHIFT dialplan function added. This function can be used to modify the
1184 pitch of a channel's tx and rx audio streams.
1185 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
1186 setting various connected line and redirecting party information.
1187 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
1188 support ISDN subaddressing.
1189 * The CHANNEL() function now supports the "name" and "checkhangup" options.
1190 * For DAHDI channels, the CHANNEL() dialplan function now allows
1191 the dialplan to request changes in the configuration of the active
1192 echo canceller on the channel (if any), for the current call only.
1195 exten => s,n,Set(CHANNEL(echocan_mode)=off)
1197 The possible values are:
1199 on - normal mode (the echo canceller is actually reinitialized)
1201 fax - FAX/data mode (NLP disabled if possible, otherwise completely
1203 voice - voice mode (returns from FAX mode, reverting the changes that
1204 were made when FAX mode was requested)
1205 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
1206 and setting variables on the channel which created the current channel.
1207 Administrators should take care to avoid naming conflicts, when multiple
1208 channels are dialled at once, especially when used with the Local channel
1209 construct (which all could set variables on the master channel). Usage
1210 of the HASH() dialplan function, with the key set to the name of the slave
1211 channel, is one approach that will avoid conflicts.
1212 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
1214 * func_odbc now allows multiple row results to be retrieved without using
1215 mode=multirow. If rowlimit is set, then additional rows may be retrieved
1216 from the same query by using the name of the function which retrieved the
1217 first row as an argument to ODBC_FETCH().
1218 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
1219 dialplan. This function returns the content of the received message.
1220 * Added REPLACE, which searches a given variable name for a set of characters,
1221 then either replaces them with a single character or deletes them.
1222 * Added PASSTHRU, which literally passes the same argument back as its return
1223 value. The intent is to be able to use a literal string argument to
1224 functions that currently require a variable name as an argument.
1225 * HASH-associated variables now can be inherited across channel creation, by
1226 prefixing the name of the hash at assignment with the appropriate number of
1227 underscores, just like variables.
1228 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
1229 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
1230 whether or not channels that are bridged to the current channel will be
1231 required to have secure signaling and/or media.
1232 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
1233 the current channel has secure signaling and/or media.
1234 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
1235 "no_media_path" option.
1236 Returns "0" if there is a B channel associated with the call.
1237 Returns "1" if no B channel is associated with the call. The call is either
1238 on hold or is a call waiting call.
1239 * Added option to dialplan function CDR(), the 'f' option
1240 allows for high resolution times for billsec and duration fields.
1241 * FILE() now supports line-mode and writing.
1242 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
1243 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
1247 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
1248 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
1249 and is set when a dynamic feature is triggered.
1250 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
1251 to dynamically create a new parking lot matching the value this varible is
1253 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
1254 features.conf that should be the base for dynamic parkinglots.
1255 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
1256 parkinglot should have.
1257 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
1258 parkinglot should have.
1259 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
1264 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
1265 timeout has expired.
1266 * Added 'R' option to app_queue. This option stops moh and indicates ringing
1267 to the caller when an Agent's phone is ringing. This can be used to indicate
1268 to the caller that their call is about to be picked up, which is nice when
1269 one has been on hold for an extened period of time.
1270 * A new config option, penaltymemberslimit, has been added to queues.conf.
1271 When set this option will disregard penalty settings when a queue has too
1273 * A new option, 'I' has been added to both app_queue and app_dial.
1274 By setting this option, Asterisk will not update the caller with
1275 connected line changes or redirecting party changes when they occur.
1276 * A 'relative-periodic-announce' option has been added to queues.conf. When
1277 enabled, this option will cause periodic announce times to be calculated
1278 from the end of announcements rather than from the beginning.
1279 * The autopause option in queues.conf can be passed a new value, "all." The
1280 result is that if a member becomes auto-paused, he will be paused in all
1281 queues for which he is a member, not just the queue that failed to reach
1283 * Added dialplan function QUEUE_EXISTS to check if a queue exists
1284 * The queue logger now allows events to optionally propagate to a file,
1285 even when realtime logging is turned on. Additionally, realtime logging
1286 supports sending the event arguments to 5 individual fields, although it
1287 will fallback to the previous data definition, if the new table layout is
1290 mISDN channel driver (chan_misdn) changes
1291 ----------------------------------------
1292 * Added display_connected parameter to misdn.conf to put a display string
1293 in the CONNECT message containing the connected name and/or number if
1294 the presentation setting permits it.
1295 * Added display_setup parameter to misdn.conf to put a display string
1296 in the SETUP message containing the caller name and/or number if the
1297 presentation setting permits it.
1298 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
1299 indicate the dialplan settings are to be obtained from the asterisk
1301 * Made misdn.conf parameter callerid accept the "name" <number> format
1302 used by the rest of the system.
1303 * Made use the nationalprefix and internationalprefix misdn.conf
1304 parameters to prefix any received number from the ISDN link if that
1305 number has the corresponding Type-Of-Number. NOTE: This includes
1306 comparing the incoming call's dialed number against the MSN list.
1307 * Added the following new parameters: unknownprefix, netspecificprefix,
1308 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
1309 received number from the ISDN link if that number has the corresponding
1311 * Added new dialplan application misdn_command which permits controlling
1312 the CCBS/CCNR functionality.
1313 * Added new dialplan function mISDN_CC which permits retrieval of various
1314 values from an active call completion record.
1315 * For PTP, you should manually send the COLR of the redirected-to party
1316 for an incomming redirected call if the incoming call could experience
1317 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
1318 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
1319 if the REDIRECTING(from-num) is not empty.
1320 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
1321 option on all of the REDIRECTING statements before dialing the
1322 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
1323 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
1324 redirecting-to presentation (COLR) when it becomes available.
1325 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
1328 thirdparty mISDN enhancements
1329 -----------------------------
1330 mISDN has been modified by Digium, Inc. to greatly expand facility message
1332 * Enhanced COLP support for call diversion and transfer.
1333 * CCBS/CCNR support.
1335 The latest modified mISDN v1.1.x based version is available at:
1336 http://svn.digium.com/svn/thirdparty/mISDN/trunk
1337 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
1339 Tagged versions of the modified mISDN code are available under:
1340 http://svn.digium.com/svn/thirdparty/mISDN/tags
1341 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
1343 libpri channel driver (chan_dahdi) DAHDI changes
1344 -------------------------------------------
1345 * The channel variable PRIREDIRECTREASON is now just a status variable
1346 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
1347 to read and alter the reason.
1348 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
1349 redirected-to party for an incomming redirected call if the incoming call
1350 could experience further redirects. Just set the
1351 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
1352 to the COLR. A call has been redirected if the REDIRECTING(count) is not
1354 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
1355 use the inhibit(i) option on all of the REDIRECTING statements before
1356 dialing the redirected-to party. You still have to set the
1357 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
1358 will update the redirecting-to presentation (COLR) when it becomes available.
1359 * Added the ability to ignore calls that are not in a Multiple Subscriber
1360 Number (MSN) list for PTMP CPE interfaces.
1361 * Added dynamic range compression support for dahdi channels. It is
1362 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
1363 * Added support for ISDN calling and called subaddress with partial support
1364 for connected line subaddress.
1365 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
1366 * Added handling of received HOLD/RETRIEVE messages and the optional ability
1367 to transfer a held call on disconnect similar to an analog phone.
1368 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
1369 Will reroute/deflect an outgoing call when receive the message.
1370 Can use the DAHDISendCallreroutingFacility to send the message for the
1372 * Added standard location to add options to chan_dahdi dialing:
1373 Dial(DAHDI/g1[/extension[/options]])
1376 R Reverse charging indication
1377 * Added Reverse Charging Indication (Collect calls) send/receive option.
1378 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
1379 Dial(DAHDI/g1/extension/R)
1380 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
1381 (requires latest LibPRI)
1382 * Added ability to send/receive keypad digits in the SETUP message.
1383 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
1384 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
1385 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
1386 (requires latest LibPRI)
1387 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
1388 to eliminate tromboned calls. A tromboned call goes out an interface and comes
1389 back into the same interface. Tromboned calls happen because of call routing,
1390 call deflection, call forwarding, and call transfer.
1391 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
1392 * Added the ability to support call waiting calls. (The SETUP has no B channel
1394 * Added Malicious Call ID (MCID) event to the AMI call event class.
1395 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
1397 Asterisk Manager Interface
1398 --------------------------
1399 * The Hangup action now accepts a Cause header which may be used to
1400 set the channel's hangup cause.
1401 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
1402 to specify a separate .pem file to hold a private key. By default sslcert
1403 is used to hold both the public and private key.
1404 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
1405 for options containing the 'tls' prefix. For example, 'sslenable' is now
1406 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
1407 across all .conf files. All affected sample.conf files have been modified to
1408 reflect this change. Previous options such as 'sslenable' still work,
1409 but options with the 'tls' prefix are preferred.
1410 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
1411 in a channel. (res_mutestream.so)
1412 * The configuration file manager.conf now supports a channelvars option, which
1413 specifies a list of channel variables to include in each channel-oriented
1415 * The redirect command now has new parameters ExtraContext, ExtraExtension,
1416 and ExtraPriority to allow redirecting the second channel to a different
1417 location than the first.
1418 * Added new event "JabberStatus" in the Jabber module to monitor buddies
1420 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
1421 in a MixMonitor recording.
1422 * The 'iax2 show peers' output is now similar to the expected output of
1424 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
1426 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
1427 AOC-E messages on a channel.
1428 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
1429 conform more closely to similar events.
1430 * Added a new eventfilter option per user to allow whitelisting and blacklisting
1432 * Added optional parkinglot variable for park command.
1433 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
1434 if CallerIDNum and CallerIDName headers are also present.
1436 Channel Event Logging
1437 ---------------------
1438 * A new interface, CEL, is introduced here. CEL logs single events, much like
1439 the AMI, but it differs from the AMI in that it logs to db backends much
1440 like CDR does; is based on the event subsystem introduced by Russell, and
1441 can share in all its benefits; allows multiple backends to operate like CDR;
1442 is specialized to event data that would be of concern to billing sytems,
1443 like CDR. Backends for logging and accounting calls have been produced,
1444 but a new CDR backend is still in development.
1448 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
1449 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
1450 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
1451 * Multiple files and formats can now be specified in cdr_custom.conf.
1452 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
1453 See configs/cdr_syslog.conf.sample for more information.
1454 * A 'sequence' field has been added to CDRs which can be combined with
1455 linkedid or uniqueid to uniquely identify a CDR.
1456 * Handling of billsec and duration field has changed. If your table definition
1457 specifies those fields as float,double or similar they will now be logged with
1458 microsecond accuracy instead of a whole integer.
1460 Calendaring for Asterisk
1461 ------------------------
1462 * A new set of modules were added supporing calendar integration with Asterisk.
1463 Dialplan functions for reading from and writing to calendars are included,
1464 as well as the ability to execute dialplan logic upon calendar event notifications.
1465 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
1466 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
1467 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
1468 2003 support does not support forms-based authentication).
1470 Call Completion Supplementary Services for Asterisk
1471 ---------------------------------------------------
1472 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
1473 DAHDI/ISDN supports call completion for the following switch types:
1474 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
1475 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
1477 Multicast RTP Support
1478 ---------------------
1479 * A new RTP engine and channel driver have been added which supports Multicast RTP.
1480 The channel driver can be used with the Page application to perform multicast RTP
1481 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
1482 Type can be either basic or linksys.
1483 Destination is the IP address and port for the RTP packets.
1484 Control address is specific to the linksys type and is used for sending the control
1485 packets unique to them.
1487 Security Events Framework
1488 -------------------------
1489 * Asterisk has a new C API for reporting security events. The module res_security_log
1490 sends these events to the "security" logger level. Currently, AMI is the only
1491 Asterisk component that reports security events. However, SIP support will be
1492 coming soon. For more information on the security events framework, see the
1493 "Asterisk Security Framework" section of the Asterisk wiki at
1494 https://wiki.asterisk.org/wiki/x/wgBQ
1495 * SIP support was added in Asterisk 10
1496 * This API now supports IPv6 addresses
1500 * A technology independent fax frontend (res_fax) has been added to Asterisk.
1501 * A spandsp based fax backend (res_fax_spandsp) has been added.
1502 * The app_fax module has been deprecated in favor of the res_fax module and
1503 the new res_fax_spandsp backend.
1504 * The SendFAX and ReceiveFAX applications now send their log messages to a
1505 'fax' logger level, instead of to the generic logger levels. To see these
1506 messages, the system's logger.conf file will need to direct the 'fax' logger
1507 level to one or more destinations; the logger.conf.sample file includes an
1508 example of how to do this. Note that if the 'fax' logger level is *not*
1509 directed to at least one destination, log messages generated by these
1510 applications will be lost, and that if the 'fax' logger level is directed to
1511 the console, the 'core set verbose' and 'core set debug' CLI commands will
1512 have no effect on whether the messages appear on the console or not.
1516 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
1517 Now, in order to enable transmitting silence during record the transmit_silence
1518 option should be used. transmit_silence_during_record remains a valid option, but
1519 defaults to the behavior of the transmit_silence option.
1520 * Addition of the Unit Test Framework API for managing registration and execution
1521 of unit tests with the purpose of verifying the operation of C functions.
1522 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
1523 XMPP text messages to the remote JID.
1524 * Modules.conf has a new option - "require" - that marks a module as critical for
1525 the execution of Asterisk.
1526 If one of the required modules fail to load, Asterisk will exit with a return
1528 * An 'X' option has been added to the asterisk application which enables #exec support.
1529 This allows #exec to be used in asterisk.conf.
1530 * jabber.conf supports a new option auth_policy that toggles auto user registration.
1531 * A new lockconfdir option has been added to asterisk.conf to protect the
1532 configuration directory (/etc/asterisk by default) during reloads.
1533 * The parkeddynamic option has been added to features.conf to enable the creation
1534 of dynamic parkinglots.
1535 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
1536 the reportalarms config option.
1537 * chan_dahdi supports dialing configuring and dialing by device file name.
1538 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
1539 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
1540 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
1541 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
1542 Handy for the above name-based syntax as it does not depend on
1543 initialization order.
1544 * The Realtime dialplan switch now caches entries for 1 second. This provides a
1545 significant increase in performance (about 3X) for installations using this switchtype.
1546 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
1547 AIS. For more information, please see the Distributed Device State section of the
1548 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1549 * The addition of G.719 pass-through support.
1550 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
1551 during device configuration.
1552 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
1553 have less than 3 lines on the LCD.
1554 * Realtime now supports database failover. See the sample extconfig.conf for details.
1555 * The addition of improved translation path building for wideband codecs. Sample
1556 rate changes during translation are now avoided unless absolutely necessary.
1557 * The addition of the res_stun_monitor module for monitoring and reacting to network
1558 changes while behind a NAT.
1559 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
1560 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
1561 These allow support for any Administration. Default is AT&T values.
1565 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
1566 optionally accept a filename, to apply the setting only to the code generated from
1567 that source file when Asterisk was built. However, there are some modules in Asterisk
1568 that are composed of multiple source files, so this did not result in the behavior
1569 that users expected. In this version, 'core set debug' and 'core set verbose'
1570 can optionally accept *module* names instead (with or without the .so extension),
1571 which applies the setting to the entire module specified, regardless of which source
1572 files it was built from.
1573 * New 'manager show settings' command showing the current settings loaded from
1575 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
1576 the channel hangup request to all channels.
1577 * Added a "core reload" CLI command that executes a global reload of Asterisk.
1579 ------------------------------------------------------------------------------
1580 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
1581 ------------------------------------------------------------------------------
1585 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
1586 Snom phones use this for call pickup of extensions that the phone is
1588 * Added support for setting the domain in the URI for caller of an
1589 outbound call by using the SIPFROMDOMAIN channel variable.
1590 * Added a new configuration option "remotesecret" for authentication to
1591 remote services. For backwards compatibility, "secret" still has the
1592 same function as before, but now you can configure both a remote secret and a
1593 local secret for mutual authentication.
1594 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
1595 the sound will be played to the target of an attended transfer
1596 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
1597 finer control over how many peers Asterisk will qualify and the gap between them
1598 when all peers need to be qualified at the same time.
1599 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
1600 (either globally or for a specific peer), chan_sip will treat any SDP data
1601 it receives as new data and update the media stream accordingly. By
1602 default, Asterisk will only modify the media stream if the SDP session
1603 version received is different from the current SDP session version. This
1604 option is required to interoperate with devices that have non-standard SDP
1605 session version implementations (observed with Microsoft OCS). This option
1606 is disabled by default.
1607 * The parsing of register => lines in sip.conf has been modified to allow a port
1608 to be present in the "user" portion. Please see the sip.conf.sample file for more
1610 * Added support for subscribing to MWI on a remote server and making the status available
1611 as a mailbox. Please see the sip.conf.sample file for more information.
1612 * Added a function to remove SIP headers added in the dialplan before the
1613 first INVITE is generated - SIPRemoveHeader()
1614 * Channel variables set with setvar= in a device configuration is now
1615 set both for inbound and outbound calls.
1616 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
1620 * Added immediate option to iax.conf
1621 * Added forceencryption option to iax.conf
1622 * Added Encryption and Trunk status to manager command "iaxpeers"
1626 * The configuration file now holds separate sections for devices and lines.
1627 Please have a look at configs/skinny.conf.sample and change your skinny.conf
1632 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
1633 support for LibOpenR2. http://www.libopenr2.org/
1634 * The UK option waitfordialtone has been added for use with BT analog
1636 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
1637 is used in conjunction with the 'faxdetect' configuration option. When
1638 'faxbuffers' is used and fax tones are detected, the channel will dynamically
1639 switch to the configured faxbuffers policy. For example, to use 6 buffers
1640 and a 'full' buffer policy for a fax transmission, add:
1642 The faxbuffers configuration will be in affect until the call is torn down.
1643 * Added service message support for 4ESS/5ESS switches.
1647 * For DAHDI channels, the CHANNEL() dialplan function now
1648 supports changing the channel's buffer policy (for the current
1649 call only), using this syntax:
1651 exten => s,n,Set(CHANNEL(buffers)=6,full)
1653 This would change the channel to the 'full' buffer policy and
1654 6 (six) buffers. Possible options for this setting are the same
1655 as those in chan_dahdi.conf.
1656 * Added a new dialplan function, CURLOPT, which permits setting various
1657 options that may be useful with the CURL dialplan function, such as
1658 cookies, proxies, connection timeouts, passwords, etc.
1659 * Permit the syntax and synopsis fields of the corresponding dialplan
1660 functions to be individually set from func_odbc.conf.
1661 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
1662 * func_odbc now may specify an insert query to execute, when the write query
1663 affects 0 rows (usually indicating that no such row exists).
1664 * Added a new dialplan function, LISTFILTER, which permits removing elements
1665 from a set list, by name. Uses the same general syntax as the existing CUT
1666 and FIELDQTY dialplan functions, which also manage lists.
1667 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
1668 obtaining realtime data from the dialplan.
1669 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
1670 a subroutine when using the GoSub() and Return() applications.
1671 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
1672 of "core show function AUDIOHOOK_INHERIT" from the CLI
1673 * Added AES_ENCRYPT. For information on its use, please see the output
1674 of "core show function AES_ENCRYPT" from the CLI
1675 * Added AES_DECRYPT. For information on its use, please see the output
1676 of "core show function AES_DECRYPT" from the CLI
1677 * func_odbc now supports database transactions across multiple queries.
1681 * Scheduled meetme conferences may now have their end times extended by
1683 * app_authenticate now gives the ability to select a prompt other than
1685 * app_directory now pays attention to the searchcontexts setting in
1686 voicemail.conf and will look through all contexts, if no context is
1687 specified in the initial argument.
1688 * A new application, Originate, has been introduced, that allows asynchronous
1689 call origination from the dialplan.
1690 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
1691 in addition to the setting in the "general" context.
1692 * Added ConfBridge dialplan application which does conference bridges without
1693 DAHDI. For information on its use, please see the output of
1694 "core show application ConfBridge" from the CLI.
1698 * The Asterisk CLI has a new command, "channel redirect", which is similar in
1699 operation to the AMI Redirect action.
1700 * extensions.conf now allows you to use keyword "same" to define an extension
1701 without actually specifying an extension. It uses exactly the same pattern
1702 as previously used on the last "exten" line. For example:
1703 exten => 123,1,NoOp(something)
1704 same => n,SomethingElse()
1705 * musiconhold.conf classes of type 'files' can now use relative directory paths,
1706 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
1707 * All deprecated CLI commands are removed from the sourcecode. They are now handled
1708 by the new clialiases module. See cli_aliases.conf.sample file.
1709 * Times within timespecs are now accurate down to the minute. This is a change
1710 from historical Asterisk, which only provided timespecs rounded to the nearest
1711 even (read: evenly divisible by 2) minute mark.
1712 * The realtime switch now supports an option flag, 'p', which disables searches for
1714 * In addition to a time range and date range, timespecs now accept a 5th optional
1715 argument, timezone. This allows you to perform time checks on alternate
1716 timezones, especially if those daylight savings time ranges vary from your
1717 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
1719 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
1720 give you the correct output for an asterisk box behind nat. It will give you the
1721 externhost and localnet settings.
1722 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
1723 can connect calls in passthrough mode, as well as record and play back files.
1724 * Successful and unsuccessful call pickup can now be alerted through sounds, by
1725 using pickupsound and pickupfailsound in features.conf.
1726 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
1727 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
1728 instead of the /var/run/asterisk.pid where it used to be. This will make
1729 installs as non-root easier to manage.
1734 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
1735 be written; they will no longer be explicitly written.
1737 Asterisk Manager Interface
1738 --------------------------
1739 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
1740 a non-empty value) in your request. If you do this, any pending AMI events will
1741 *not* be included in the response to your request as they would normally, but
1742 will be left in the event queue for the next request you make to retrieve. For
1743 some applications, this will allow you to guarantee that you will only see
1744 events in responses to 'WaitEvent' actions, and can better know when to expect them.
1745 To know whether the Asterisk server supports this header or not, your client can
1746 inspect the first response back from the server to see if it includes this header:
1748 Pragma: SuppressEvents
1750 If this is included, the server supports event suppression.
1752 * Added 4 new Actions to list skinny device(s) and line(s)
1758 LDAP Schema File Additions
1759 --------------------------
1760 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
1761 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
1763 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
1764 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
1765 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
1766 * Removed redundant IPaddr (there's already IPAddress)
1767 - Gives more configuration Flags for SIP-Users available (tested)
1768 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
1769 without extensibleObject (which really should be the last resort); gives
1770 also additional possibilities for LDAP-filter
1772 ------------------------------------------------------------------------------
1773 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
1774 ------------------------------------------------------------------------------
1776 Device State Handling
1777 ---------------------
1778 * The event infrastructure in Asterisk got another big update to help support
1779 distributed events. It currently supports distributed device state and
1780 distributed Voicemail MWI (Message Waiting Indication). A new module has
1781 been merged, res_ais, which facilitates communicating events between servers.
1782 It uses the SAForum AIS (Service Availability Forum Application Interface
1783 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
1784 a cluster of Asterisk servers, and to share events between them. For more
1785 information on setting this up, refer to the Distributed Device State section
1786 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1790 * Added a new dialplan function, AST_CONFIG(), which allows you to access
1791 variables from an Asterisk configuration file.
1792 * The JACK_HOOK function now has a c() option to supply a custom client name.
1793 * Added two new dialplan functions from libspeex for audio gain control and
1794 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
1795 rx directions of a channel from the dialplan.
1796 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
1797 based on other parameters. The default is still to search based on the
1798 forwarding station ID. However, there are new options that allow you to search
1799 based on the message desk terminal ID, or the message desk number.
1800 * TIMEOUT() has been modified to be accurate down to the millisecond.
1801 * ENUM*() functions now include the following new options:
1802 - 'u' returns the full URI and does not strip off the URI-scheme.
1803 - 's' triggers ISN specific rewriting
1804 - 'i' looks for branches into an Infrastructure ENUM tree
1805 - 'd' for a direct DNS lookup without any flipping of digits.
1806 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
1807 * CHANNEL() now has options for the maximum, minimum, and standard or normal
1808 deviation of jitter, rtt, and loss for a call using chan_sip.
1810 DAHDI channel driver (chan_dahdi) Changes
1811 ----------------------------------------
1812 * Channels can now be configured using named sections in chan_dahdi.conf, just
1813 like other channel drivers, including the use of templates.
1814 * The default for pridialplan has changed from 'national' to 'unknown'.
1818 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
1819 to something that matches the pattern a hint will be created using the contents
1820 and variables evaluated.
1821 * Dialplan matching has been extended to allow an extension to return to the
1822 PBX core to wait for more digits. This is done by using the new dialplan
1823 application called "Incomplete". This will permit a whole new level of
1824 extension control, by giving the administrator more control over early
1825 matches employing one of the short-circuit pattern match operators. Note
1826 that custom applications can trigger this same behavior by returning the
1827 special value AST_PBX_INCOMPLETE.
1831 * Directory now permits both first and last names to be matched at the same
1832 time. In addition, the number of digits to enter of the name can be set in
1833 the arguments to Directory; previously, you could enter only 3, regardless
1834 of how many names are in your company. For large companies, this should be
1836 * Voicemail now permits a mailbox setting to wrap around from first to last
1837 messages, if the "messagewrap" option is set to a true value.
1838 * Voicemail now permits an external script to be run, for password validation.
1839 The script should output "VALID" or "INVALID" on stdout, depending upon the
1840 wish to validate or invalidate the password given. Arguments are:
1841 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
1843 * Dial has a new option: F(context^extension^pri), which permits a callee to
1844 continue in the dialplan, at the specified label, if the caller hangs up.
1845 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
1846 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
1847 * The Jack application now has a c() option to supply a custom client name.
1848 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
1849 like the pre-existing whisper mode, except that the spy can also talk to the
1850 participant on the bridged channel as well.
1851 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
1852 to be spoken instead of the channel name or number. For more information on the
1853 use of this option, issue the command "core show application ChanSpy" from the
1855 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
1856 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
1857 words, if using the 'd' option, it is not possible to enter a number to append to
1858 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
1859 change to whisper mode, and pressing 6 will change to barge mode.
1860 * ExternalIVR now takes several options that affect the way it performs, as
1861 well as having several new commands. Please see the External IVR page on the Asterisk
1862 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
1863 * Added ability to communicate over a TCP socket instead of forking a child process for the
1864 ExternalIVR application.
1865 * ChanIsAvail has a new option, 'a', which will return all available channels instead
1866 of just the first one if you give the function more then one channel to check.
1867 * PrivacyManager now takes an option where you can specify a context where the
1868 given number will be matched. This way you have more control over who is allowed
1869 and it stops the people who blindly enter 10 digits.
1870 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
1871 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
1872 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
1873 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
1874 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
1875 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
1876 * The Dial() application no longer copies the language used by the caller to the callee's
1877 channel. If you desire for the caller's channel's language to be used for file playback
1878 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
1879 * SendImage() no longer hangs up the channel on error; instead, it sets the
1880 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
1881 'UNSUPPORTED'. This change makes SendImage() more consistent with other
1883 * Park has a new option, 's', which silences the announcement of the parking space number.
1884 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
1885 invalid input and will be assumed to mean that no timeout is desired.
1889 * Added DNS manager support to registrations for peers referencing peer entries.
1890 DNS manager runs in the background which allows DNS lookups to be run asynchronously
1891 as well as periodically updating the IP address. These properties allow for
1892 better performance as well as recovery in the event of an IP change.
1893 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
1894 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
1895 These changes also provide performance improvements for call setup and tear down.
1896 * Added ability to specify registration expiry time on a per registration basis in
1898 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
1900 * Added t38pt_usertpsource option. See sip.conf.sample for details.
1901 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
1902 * 'sip show peers' and 'sip show users' display their entries sorted in
1903 alphabetical order, as opposed to the order they were in, in the config
1905 * Videosupport now supports an additional option, "always", which always sets
1906 up video RTP ports, even on clients that don't support it. This helps with
1907 callfiles and certain transfers to ensure that if two video phones are
1908 connected, they will always share video feeds.
1912 * Existing DNS manager lookups extended to check for SRV records.
1913 * IAX2 encryption support has been improved to support periodic key rotation
1914 within a call for enhanced security. The option "keyrotate" has been
1915 provided to disable this functionality to preserve backwards compatibility
1916 with older versions of IAX2 that do not support key rotation.
1920 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
1921 data tree based on the given <path>.
1922 * New CLI command "data show providers" that will display all the registered
1924 * New CLI command, "config reload <file.conf>" which reloads any module that
1925 references that particular configuration file. Also added "config list"
1926 which shows which configuration files are in use.
1927 * New CLI commands, "pri show version" and "ss7 show version" that will
1928 display which version of libpri and libss7 are being used, respectively.
1929 A new API call was added so trunk will now have to be compiled against
1930 a versions of libpri and libss7 that have them or it will not know that
1931 these libraries exist.
1932 * The commands "core show globals", "core set global" and "core set chanvar" has
1933 been deprecated in favor of the more semanticly correct "dialplan show globals",
1934 "dialplan set chanvar" and "dialplan set global".
1935 * New CLI command "dialplan show chanvar" to list all variables associated
1936 with a given channel.
1940 * Addresses managed by DNS manager now can check to see if there is a DNS
1941 SRV record for a given domain and will use that hostname/port if present.
1943 AMI - The manager (TCP/TLS/HTTP)
1944 --------------------------------
1945 * The Status command now takes an optional list of variables to display
1946 along with channel status.
1947 * The QueueEntry event now also includes the channel's uniqueid
1951 * res_odbc no longer has a limit of 1023 total possible unshared connections,
1952 as some people were running into this limit. This limit has been increased
1957 * The TRANSFER queue log entry now includes the the caller's original
1958 position in the transferred-from queue.
1959 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
1960 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
1961 as well as an explanation about timeout options in general
1962 * Added a new option - C - for forcing the "answered elsewhere" flag on
1963 cancellation of calls in to members of the queue. This is to avoid the
1964 call to a member of a queue having the call listed as a "missed call".
1968 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
1969 adaptive capabilities. What this means in practical terms is that if your
1970 realtime table lacks critical fields, Asterisk will now emit warnings to
1971 that effect. Also, some of the realtime drivers have the ability (if
1972 configured) to automatically add those columns to the table with the
1973 correct type and length.
1977 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
1978 the 'setvar' option to cause a given audio file to be played upon completion
1979 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
1980 Skinny channels only.
1981 * You can now compile Asterisk against the Hoard Memory Allocator, see the
1982 Hoard page on the Asterisk wiki for more information:
1983 https://wiki.asterisk.org/wiki/x/pQBB
1984 * Config file variables may now be appended to, by using the '+=' append
1985 operator. This is most helpful when working with long SQL queries in
1986 func_odbc.conf, as the queries no longer need to be specified on a single
1988 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
1989 which will add a second to the billsec when the ending
1990 time is set, if the number in the microseconds field of the end time is
1991 greater than the number of microseconds in the answer time. This allows
1992 users to count the 'initiated' seconds in their billing records.
1994 ------------------------------------------------------------------------------
1995 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
1996 ------------------------------------------------------------------------------
1998 AMI - The manager (TCP/TLS/HTTP)
1999 --------------------------------
2000 * Manager has undergone a lot of changes, all of them documented
2001 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
2002 * Manager version has changed to 1.1
2003 * Added a new action 'CoreShowChannels' to list currently defined channels
2004 and some information about them.
2005 * Added a new action 'SIPshowregistry' to list SIP registrations.
2006 * Added TLS support for the manager interface and HTTP server
2007 * Added the URI redirect option for the built-in HTTP server
2008 * The output of CallerID in Manager events is now more consistent.
2009 CallerIDNum is used for number and CallerIDName for name.
2010 * Enable https support for builtin web server.
2011 See configs/http.conf.sample for details.
2012 * Added a new action, GetConfigJSON, which can return the contents of an
2013 Asterisk configuration file in JSON format. This is intended to help
2014 improve the performance of AJAX applications using the manager interface
2016 * SIP and IAX manager events now use "ChannelType" in all cases where we
2017 indicate channel driver. Previously, we used a mixture of "Channel"
2018 and "ChannelDriver" headers.
2019 * Added a "Bridge" action which allows you to bridge any two channels that
2020 are currently active on the system.
2021 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
2022 the voicemail users setup.
2023 * Added 'DBDel' and 'DBDelTree' manager commands.
2024 * cdr_manager now reports events via the "cdr" level, separating it from
2025 the very verbose "call" level.
2026 * Manager users are now stored in memory. If you change the manager account
2027 list (delete or add accounts) you need to reload manager.
2028 * Added Masquerade manager event for when a masquerade happens between
2030 * Added "manager reload" command for the CLI
2031 * Lots of commands that only provided information are now allowed under the
2032 Reporting privilege, instead of only under Call or System.
2033 * The IAX* commands now require either System or Reporting privilege, to
2034 mirror the privileges of the SIP* commands.
2035 * Added ability to retrieve list of categories in a config file.
2036 * Added ability to retrieve the content of a particular category.
2037 * Added ability to empty a context.
2038 * Created new action to create a new file.
2039 * Updated delete action to allow deletion by line number with respect to category.
2040 * Added new action insert to add new variable to category at specified line.
2041 * Updated action newcat to allow new category to be inserted in file above another
2043 * Added new event "JitterBufStats" in the IAX2 channel
2044 * Originate now requires the Originate privilege and, if you want to call out
2045 to a subshell, it requires the System privilege, as well. This was done to
2046 enhance manager security.
2047 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
2048 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
2049 or manager show command Atxfer from the CLI
2050 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
2051 details or manager show command IAXregistry from the CLI
2055 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
2056 state in the dialplan, as well as creating custom device states that are
2057 controllable from the dialplan.
2058 * Extend CALLERID() function with "pres" and "ton" parameters to
2059 fetch string representation of calling number presentation indicator
2060 and numeric representation of type of calling number value.
2061 * MailboxExists converted to dialplan function
2062 * A new option to Dial() for telling IP phones not to count the call
2063 as "missed" when dial times out and cancels.
2064 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
2065 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
2066 held for any given channel. Also, locks are automatically freed when a
2068 * Added HINT() dialplan function that allows retrieving hint information.
2069 Hints are mappings between extensions and devices for the sake of
2070 determining the state of an extension. This function can retrieve the list
2071 of devices or the name associated with a hint.
2072 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
2074 * Added SYSINFO() dialplan function which allows retrieval of system information
2075 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
2076 the existence of a dialplan target.
2077 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
2078 upper and lower case, respectively.
2079 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
2080 ID for the call (not the Asterisk call ID or unique ID), provided that the
2081 channel driver supports this. For SIP, you get the SIP call-ID for the
2082 bridged channel which you can store in the CDR with a custom field.
2086 * Added CLI permissions, config file: cli_permissions.conf
2087 default is to allow all commands for every local user/group.
2088 Also this new feature added three new CLI commands:
2089 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
2090 - cli reload permissions
2091 - cli show permissions
2092 * New CLI command "core show hint" (usage: core show hint <exten>)
2093 * New CLI command "core show settings"
2094 * Added 'core show channels count' CLI command.
2095 * Added the ability to set the core debug and verbose values on a per-file basis.
2096 * Added 'queue pause member' and 'queue unpause member' CLI commands
2097 * Ability to set process limits ("ulimit") without restarting Asterisk
2098 * Enhanced "agi debug" to print the channel name as a prefix to the debug
2099 output to make debugging on busy systems much easier.
2100 * New CLI commands "dialplan set extenpatternmatching true/false"
2101 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
2102 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
2103 listed in the startup_commands section of cli.conf will get executed.
2104 * Added a CLI command, "devstate change", which allows you to set custom device
2105 states from the func_devstate module that provides the DEVICE_STATE() function
2106 and handling of the "Custom:" devices.
2107 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
2108 sorted into the different possible callbacks, with the number of entries
2109 currently scheduled for each. Gives you a feel for how busy the sip channel
2111 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
2112 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
2113 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
2117 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
2118 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
2119 for a received call. If it is detected, the channel will jump to the
2120 'fax' extension in the dialplan.
2121 * The default SIP useragent= identifier now includes the Asterisk version
2122 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
2123 If set, and the incoming request carries authentication info,
2124 the username to match in the users list is taken from the Digest header
2125 rather than from the From: field. This feature is considered experimental.
2126 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
2127 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
2128 * The "localmask" setting was removed in version 1.2 and the reminder about it
2129 being removed is now also removed.
2130 * A new option "busylevel" for setting a level of calls where asterisk reports
2131 a device as busy, to separate it from call-limit. This value is also added
2132 to the SIP_PEER dialplan function.
2133 * A new realtime family called "sipregs" is now supported to store SIP registration
2134 data. If this family is defined, "sippeers" will be used for configuration and
2135 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
2136 registration data, as before.
2137 * The SIPPEER function have new options for port address, call and pickup groups
2138 * Added support for T.140 realtime text in SIP/RTP
2139 * The "checkmwi" option has been removed from sip.conf, as it is no longer
2140 required due to the restructuring of how MWI is handled. See the descriptions
2141 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
2142 for more information.
2143 * Added rtpdest option to CHANNEL() dialplan function.
2144 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
2145 * SIP now adds a header to the CANCEL if the call was answered by another phone
2146 in the same dial command, or if the new c option in dial() is used.
2147 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
2148 states it is not needed. For phones, however, that do require it the "registertrying" option
2149 has been added so it can be enabled.
2150 * A new option called "callcounter" (global/peer/user level) enables call counters needed
2151 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
2152 used to enable this functionality).
2153 * New settings for timer T1 and timer B on a global level or per device. This makes it
2154 possible to force timeout faster on non-responsive SIP servers. These settings are
2155 considered advanced, so don't use them unless you have a problem.
2156 * Added a dial string option to be able to set the To: header in an INVITE to any
2158 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
2159 the qualify frequency.
2160 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
2161 were not properly torn down due to network or endpoint failures during an established
2163 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
2164 and configs/sip.conf.sample for more information on how it is used.
2165 * Added a new configuration option "authfailureevents" that enables manager events when
2166 a peer can't authenticate properly.
2167 * Added DNS manager support to registrations for peers not referencing a peer entry.
2171 * Added the trunkmaxsize configuration option to chan_iax2.
2172 * Added the srvlookup option to iax.conf
2173 * Added support for OSP. The token is set and retrieved through the CHANNEL()
2176 XMPP Google Talk/Jingle changes
2177 -------------------------------
2178 * Added the bindaddr option to gtalk.conf.
2182 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
2183 * Proper codec support in chan_skinny.
2184 * Added settings for IP and Ethernet QoS requests
2188 * Added separate settings for media QoS in mgcp.conf
2190 Console Channel Driver changes
2191 ------------------------------
2192 * Added experimental support for video send & receive to chan_oss.
2193 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
2196 Phone channel changes (chan_phone)
2197 ----------------------------------
2198 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
2200 H.323 channel Changes
2201 ---------------------
2202 * H323 remote hold notification support added (by NOTIFY message
2203 and/or H.450 supplementary service)
2205 Local channel changes
2206 ---------------------
2207 * The device state functionality in the Local channel driver has been updated
2208 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
2209 to just UNKNOWN if the extension exists.
2210 * Added jitterbuffer support for chan_local. This allows you to use the
2211 generic jitterbuffer on incoming calls going to Asterisk applications.
2212 For example, this would allow you to use a jitterbuffer for an incoming
2213 SIP call to Voicemail by putting a Local channel in the middle. This
2214 feature is enabled by using the 'j' option in the Dial string to the Local
2215 channel in conjunction with the existing 'n' option for local channels.
2216 * A 'b' option has been added which causes chan_local to return the actual channel
2217 that is behind it when queried. This is useful for transfer scenarios as the
2218 actual channel will be transferred, not the Local channel.
2220 Agent channel changes
2221 ----------------------
2222 * The ackcall and endcall options are now supplemented with options acceptdtmf
2223 and enddtmf. These allow for the DTMF keypress to be configurable. The options
2224 default to their old hard-coded values ('#' and '*' respectively) so this should
2225 not break any existing agent installations.
2227 DAHDI channel driver (chan_dahdi) Changes
2228 ----------------------------------------
2229 * SS7 support (via libss7 library)
2230 * In India, some carriers transmit CID via dtmf. Some code has been added
2231 that will handle some situations. The cidstart=polarity_IN choice has been added for
2232 those carriers that transmit CID via dtmf after a polarity change.
2233 * CID matching information is now shown when doing 'dialplan show'.
2234 * Added dahdi show version CLI command.
2235 * Added setvar support to chan_dahdi.conf channel entries.
2236 * Added two new options: mwimonitor and mwimonitornotify. These options allow
2237 you to enable MWI monitoring on FXO lines. When the MWI state changes,
2238 the script specified in the mwimonitornotify option is executed. An internal
2239 event indicating the new state of the mailbox is also generated, so that
2240 the normal MWI facilities in Asterisk work as usual.
2241 * Added signalling type 'auto', which attempts to use the same signalling type
2242 for a channel as configured in DAHDI. This is primarily designed for analog
2243 ports, but will also work for digital ports that are configured for FXS or FXO
2244 signalling types. This mode is also the default now, so if your chan_dahdi.conf
2245 does not specify signalling for a channel (which is unlikely as the sample
2246 configuration file has always recommended specifying it for every channel) then
2247 the 'auto' mode will be used for that channel if possible.
2248 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
2249 state for a channel; also ensured that the DNDState Manager event is
2250 emitted no matter how the DND state is set or cleared.
2254 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
2255 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
2256 for details. This new channel driver allows you to use Nortel i2002,
2257 i2004, and i2050 phones with Asterisk.
2258 * Added a new channel driver, chan_console, which uses portaudio as a cross
2259 platform audio interface. It was written as a channel driver that would
2260 work with Mac CoreAudio, but portaudio supports a number of other audio
2261 interfaces, as well. Note that this channel driver requires v19 or higher
2262 of portaudio; older versions have a different API.
2266 * Added the ability to specify arguments to the Dial application when using
2267 the DUNDi switch in the dialplan.
2268 * Added the ability to set weights for responses dynamically. This can be
2269 done using a global variable or a dialplan function. Using the SHELL()
2270 function would allow you to have an external script set the weight for
2272 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
2273 functions will allow you to initiate a DUNDi query from the dialplan,
2274 find out how many results there are, and access each one.
2275 * Added the ability to specifiy a port for a dundi peer.
2279 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
2280 functions will allow you to initiate an ENUM lookup from the dialplan,
2281 and Asterisk will cache the results. ENUMRESULT can be used to access
2282 the results without doing multiple DNS queries.
2286 * Added the ability to customize which sound files are used for some of the
2287 prompts within the Voicemail application by changing them in voicemail.conf
2288 * Added the ability for the "voicemail show users" CLI command to show users
2289 configured by the dynamic realtime configuration method.
2290 * MWI (Message Waiting Indication) handling has been significantly
2291 restructured internally to Asterisk. It is now totally event based
2292 instead of polling based. The voicemail application will notify other
2293 modules that have subscribed to MWI events when something in the mailbox
2295 This also means that if any other entity outside of Asterisk is changing
2296 the contents of mailboxes, then the voicemail application still needs to
2297 poll for changes. Examples of situations that would require this option
2298 are web interfaces to voicemail or an email client in the case of using
2299 IMAP storage. So, two new options have been added to voicemail.conf
2300 to account for this: "pollmailboxes" and "pollfreq". See the sample
2301 configuration file for details.
2302 * Added "tw" language support
2303 * Added support for storage of greetings using an IMAP server
2304 * Added ability to customize forward, reverse, stop, and pause keys for message playback
2305 * SMDI is now enabled in voicemail using the smdienable option.
2306 * A "lockmode" option has been added to asterisk.conf to configure the file
2307 locking method used for voicemail, and potentially other things in the
2308 future. The default is the old behavior, lockfile. However, there is a
2309 new method, "flock", that uses a different method for situations where the
2310 lockfile will not work, such as on SMB/CIFS mounts.
2311 * Added the ability to backup deleted messages, to ease recovery in the case
2312 that a user accidentally deletes a message, and discovers that they need it.
2313 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
2314 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
2315 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
2316 voicemail boxes. The SMDI interface can also poll for MWI changes when some
2317 outside entity is modifying the state of the mailbox (such as IMAP storage or
2318 a web interface of some kind).
2319 * Added the support for marking messages as "urgent." There are two methods to accomplish
2320 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
2321 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
2322 the message as urgent after he has recorded a voicemail by following the voice instructions.
2323 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
2328 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
2329 used across multiple queues.
2330 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
2331 setqueueentryvar options for each queue, see queues.conf.sample for details.
2332 * Added keepstats option to queues.conf which will keep queue
2333 statistics during a reload.
2334 * setinterfacevar option in queues.conf also now sets a variable
2335 called MEMBERNAME which contains the member's name.
2336 * Added 'Strategy' field to manager event QueueParams which represents
2337 the queue strategy in use.
2338 * Added option to run macro when a queue member is connected to a caller,
2339 see queues.conf.sample for details.
2340 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
2341 does not count paused queue members as unavailable.
2342 * Added min-announce-frequency option to queues.conf which allows you to control the
2343 minimum amount of time between queue announcements for use when the caller's queue
2344 position changes frequently.
2345 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
2347 * Added ability for non-realtime queues to have realtime members
2348 * Added the "linear" strategy to queues.
2349 * Added the "wrandom" strategy to queues.
2350 * Added new channel variable QUEUE_MIN_PENALTY
2351 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
2352 rules in queuerules.conf. See configs/queuerules.conf.sample for details
2353 * Added a new parameter for member definition, called state_interface. This may be
2354 used so that a member may be called via one interface but have a different interface's
2355 device state reported.
2356 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
2357 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
2358 "manager show command QueueReset."
2359 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
2360 specified by the periodic-announce option, then one will be chosen randomly when it is time
2361 to play a periodic announcment
2362 * New configuration options: announce-position now takes two more values in addition to "yes" and
2363 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
2364 announce-position-limit. By setting announce-position to "limit" callers will only have their
2365 position announced if their position is less than what is specified by announce-position-limit.
2366 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
2367 will be told that their are more than announce-position-limit callers waiting.
2368 * Two new queue log events have been added. An ADDMEMBER event will be logged
2369 when a realtime queue member is added and a REMOVEMEMBER event will be logged
2370 when a realtime queue member is removed. Since there is no calling channel associated
2371 with these events, the string "REALTIME" is placed where the channel's unique id
2372 is typically placed.
2373 * The configuration method for the "joinempty" and "leavewhenempty" options has
2374 changed to a comma-separated list of methods of determining member availability
2375 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
2376 values are still accepted for backwards-compatibility, though.
2377 * The average talktime is now calculated on queues. This information is reported via the
2378 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
2379 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
2384 * The 'o' option to provide an optimization has been removed and its functionality
2385 has been enabled by default.
2386 * When a conference is created, the UNIQUEID of the channel that caused it to be
2387 created is stored. Then, every channel that joins the conference will have the
2388 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
2389 callers that come and go from long standing conferences.
2390 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
2391 except it does operations on a channel by name, instead of number in a conference.
2392 This is a very useful feature in combination with the 'X' option to ChanSpy.
2393 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
2395 * Added new RealTime functionality to provide support for scheduled conferencing.
2396 This includes optional messages to the caller if they attempt to join before
2397 the schedule start time, or to allow the caller to join the conference early.
2398 Also included is optional support for limiting the number of callers per
2399 RealTime conference.
2400 * Added the S() and L() options to the MeetMe application. These are pretty
2401 much identical to the S() and L() options to Dial(). They let you set
2402 timeouts for the conference, as well as have warning sounds played to
2403 let the caller know how much time is left, and when it is running out.
2404 * Added the ability to do "meetme concise" with the "meetme" CLI command.
2405 This extends the concise capabilities of this CLI command to include
2406 listing all conferences, instead of an addition to the other sub commands
2407 for the "meetme" command.
2408 * Added the ability to specify the music on hold class used to play into the
2409 conference when there is only one member and the M option is used.
2410 * Added MEETME_INFO dialplan function which provides a way to query
2411 various properties of a Meetme conference.
2412 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
2413 and *84: record in-conf
2415 Other Dialplan Application Changes
2416 ----------------------------------
2417 * Argument support for Gosub application
2418 * From the to-do lists: straighten out the app timeout args:
2419 Wait() app now really does 0.3 seconds- was truncating arg to an int.
2420 WaitExten() same as Wait().
2421 Congestion() - Now takes floating pt. argument.
2422 Busy() - now takes floating pt. argument.
2423 Read() - timeout now can be floating pt.
2424 WaitForRing() now takes floating pt timeout arg.
2425 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
2426 * Added 's' option to Page application.
2427 * Added an optional timeout argument to the Page application.
2428 * Added 'E', 'V', and 'P' commands to ExternalIVR.
2429 * Added 'o' and 'X' options to Chanspy.
2430 * Added a new dialplan application, Bridge, which allows you to bridge the
2431 calling channel to any other active channel on the system.
2432 * Added the ability to specify a music on hold class to play instead of ringing
2433 for the SLATrunk application.
2434 * The Read application no longer exits the dialplan on error. Instead, it sets
2435 READSTATUS to ERROR, which you can catch and handle separately.
2436 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
2437 of asking for verification of each name, one at a time.
2438 * Privacy() no longer uses privacy.conf, as all options are specifyable as
2439 direct options to the app.
2440 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
2442 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
2443 * The ChannelRedirect application no longer exits the dialplan if the given channel
2444 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
2445 or NOCHANNEL if the given channel was not found.
2446 * The silencethreshold setting that was previously configurable in multiple
2447 applications is now settable globally via dsp.conf.
2449 Music On Hold Changes
2450 ---------------------
2451 * A new option, "digit", has been added for music on hold classes in
2452 musiconhold.conf. If this is set for a music on hold class, a caller
2453 listening to music on hold can press this digit to switch to listening
2454 to this music on hold class.
2455 * Support for realtime music on hold has been added.
2456 * In conjunction with the realtime music on hold, a general section has
2457 been added to musiconhold.conf, its sole variable is cachertclasses. If this
2458 is set, then music on hold classes found in realtime will be cached in memory.
2462 * AEL upgraded to use the Gosub with Arguments instead
2463 of Macro application, to hopefully reduce the problems
2464 seen with the artificially low stack ceiling that
2465 Macro bumps into. Macros can only call other Macros
2466 to a depth of 7. Tests run using gosub, show depths
2467 limited only by virtual memory. A small test demonstrated
2468 recursive call depths of 100,000 without problems.
2469 -- in addition to this, all apps that allowed a macro
2470 to be called, as in Dial, queues, etc, are now allowing
2471 a gosub call in similar fashion.
2472 * AEL now generates LOCAL(argname) declarations when it
2473 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
2474 etc. That makes the arguments local in scope. The user
2475 can define their own local variables in macros, now,
2476 by saying "local myvar=someval;" or using Set() in this
2477 fashion: Set(LOCAL(myvar)=someval); ("local" is now
2479 * utils/conf2ael introduced. Will convert an extensions.conf
2480 file into extensions.ael. Very crude and unfinished, but
2481 will be improved as time goes by. Should be useful for a
2482 first pass at conversion.
2483 * aelparse will now read extensions.conf to see if a referenced
2484 macro or context is there before issueing a warning.
2485 * AEL parser sets a local channel variable ~~EXTEN~~, to
2486 preserve the value of ${EXTEN} thru switch statements.
2487 * New operator in $[...] expressions: the ~~ operator serves
2488 as a concatenation operator. AT THE MOMENT, it is really only
2489 necessary and useful in AEL, especially in if() expressions.
2490 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
2491 any enclosing double-quotes, and evaluate to the value of a
2492 concatenated with the value of b. For example if a is set to
2493 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
2494 evaluate to xyzabc .
2497 Call Features (res_features) Changes
2498 ------------------------------------
2499 * Added the parkedcalltransfers option to features.conf
2500 * Added parkedcallparking option to control one touch parking w/ parking
2502 * Added parkedcallhangup option to control disconnect feature w/ parking
2504 * Added parkedcallrecording option to control one-touch record w/ parking
2506 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
2507 parkedcalltransfers option support for multiple parking lots.
2508 * Added BRIDGE_FEATURES variable to set available features for a channel
2509 * The built-in method for doing attended transfers has been updated to
2510 include some new options that allow you to have the transferee sent
2511 back to the person that did the transfer if the transfer is not successful.
2512 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
2513 in features.conf.sample.
2514 * Added support for configuring named groups of custom call features in
2515 features.conf. This means that features can be written a single time, and
2516 then mapped into groups of features for different key mappings or easier
2518 * Updated the ParkedCall application to allow you to not specify a parking
2519 extension. If you don't specify a parking space to pick up, it will grab
2520 the first one available.
2521 * Added cli command 'features reload' to reload call features from features.conf
2522 * Moved into core asterisk binary.
2523 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
2524 * Added the ability for custom parking lots to be configured with their own
2525 parking extension with the parkext option.
2527 Language Support Changes
2528 ------------------------
2529 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
2530 * Added support for the Hungarian language for saying numbers, dates, and times.
2534 * Added SPEECH commands for speech recognition. A complete listing can be found
2536 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
2537 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
2538 does not behave as expected; the native command needs to be used, instead.
2539 * Added the ability to perform SRV lookups on fast AGI calls. To use this
2540 feature, simply use hagi: instead of agi: as the protocol portion
2541 of the URI parameter to the AGI function call in your dial plan. Also note
2542 that specifying a port number in the AGI URI will disable SRV lookups,
2543 even if you use the hagi: protocol.
2544 * No longer support MSG_OOB flag on HANGUP.
2548 * Added rotatestrategy option to logger.conf, along with two new options:
2549 "timestamp" which will use the time to name the logger files instead of
2550 sequence number; and "rotate", which rotates the names of the log files,
2551 similar to the way syslog rotates files.
2552 * Added exec_after_rotate option to logger.conf, which allows a system
2553 command to be run after rotation. This is primarily useful with
2554 rotatestrategy=rotate, to allow a limit on the number of log files kept
2555 and to ensure that the oldest log file gets deleted.
2556 * Added realtime support for the queue log
2560 * The cdr_manager module has a [mappings] feature, like cdr_custom,
2561 to add fields to the manager event from the CDR variables.
2562 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
2563 backend database CDR table. Specifically, additional, non-standard
2564 columns are supported, merely by setting the corresponding CDR variable in
2565 your dialplan. In addition, you may alias any column to another name (for
2566 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
2567 simply "alias src => ANI" in the configuration file). Records may be
2568 posted to more than one backend, simply by specifying multiple categories
2569 in the configuration file. And finally, you may filter which CDRs get
2570 posted to each backend, by specifying a filter (which the record must
2571 match) for the particular category. Filters are additive (meaning all
2572 rules must match to post that CDR).
2573 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
2574 module. Specifically, you may add additional columns into the table and
2575 they will be set, if you set the corresponding CDR variable name. Also,
2576 if you omit columns in your database table, they will be silently skipped
2577 (but a record will still be inserted, based on what columns remain). Note
2578 that the other two features from cdr_adaptive_odbc (alias and filter) are
2579 not currently supported.
2580 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
2581 has been disabled using the NoCDR application.
2583 Miscellaneous New Modules
2584 -------------------------
2585 * Added a new CDR module, cdr_sqlite3_custom.
2586 * Added a new realtime configuration module, res_config_sqlite
2587 * Added a new codec translation module, codec_resample, which re-samples
2588 signed linear audio between 8 kHz and 16 kHz to help support wideband
2590 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
2591 based on configuration templates that use Asterisk dialplan function and
2592 variable substitution. It should be possible to create phone profiles and
2593 templates that work for the majority of phones provisioned over http. It
2594 is currently only intended to provision a single user account per phone.
2595 An example profile and set of templates for Polycom phones is provided.
2596 NOTE: Polycom firmware is not included, but should be placed in
2597 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
2598 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
2599 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
2600 provided; there is a JACK() application, and a JACK_HOOK() function. Both
2601 interfaces create an input and output JACK port. The application makes
2602 these ports the endpoint of the call. The audio coming from the channel
2603 goes out the output port and whatever comes back in on the input port is
2604 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
2605 audiohook on the channel. This lets you run the audio coming from a
2606 channel through JACK, and whatever comes back in is what gets forwarded
2607 on as the channel's audio. This is very useful for building custom
2608 vocoders or doing recording or analysis of the channel's audio in another
2610 * Added a new module, res_config_curl, which permits using a HTTP POST url
2611 to retrieve, create, update, and delete realtime information from a remote
2612 web server. Note that this module requires func_curl.so to be loaded for
2613 backend functionality.
2614 * Added a new module, res_config_ldap, which permits the use of an LDAP
2615 server for realtime data access.
2616 * Added support for writing and running your dialplan in lua using the pbx_lua
2617 module. See configs/extensions.lua.sample for examples of how to do this.
2621 * Ability to use libcap to set high ToS bits when non-root
2622 on Linux. If configure is unable to find libcap then you
2623 can use --with-cap to specify the path.
2624 * Added maxfiles option to options section of asterisk.conf which allows you to specify
2625 what Asterisk should set as the maximum number of open files when it loads.
2626 * Added the jittertargetextra configuration option.
2627 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
2628 configuration files for the IP channel drivers. The new option is "cos".
2629 This information is also documented on the Asterisk wiki at
2630 https://wiki.asterisk.org/wiki/x/EYBG
2631 * When originating a call using AMI or pbx_spool that fails the reason for failure
2632 will now be available in the failed extension using the REASON dialplan variable.
2633 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
2634 It allows you to configure a prefix for auto-monitor recordings.
2635 * A new extension pattern matching algorithm, based on a trie, is introduced
2636 here, that could noticeably speed up mid-sized to large dialplans.
2637 It is NOT used by default, as duplicating the behaviour of the old pattern
2638 matcher is still under development. A config file option, in extensions.conf,
2639 in the [general] section, called "extenpatternmatchingnew", is by default
2640 set to false; setting that to true will force the use of the new algorithm.
2641 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
2642 be used to switch the algorithms at run time.
2643 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
2644 specifying which socket to use to connect to the running Asterisk daemon
2646 * Performance enhancements to the sched facility, which is used in
2647 the channel drivers, etc. Added hashtabs and doubly-linked lists
2648 to speed up deletion; start at the beginning or end of list to
2650 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
2651 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
2652 Added regression tests to the tests/ dir, also.
2653 * Added a refcount trace feature to astobj2 for those trying to balance
2654 object creation, deletion; work, play; space and time. See the
2655 notes in astobj2.h. Also, see utils/refcounter as well, as a
2656 quick way to find unbalanced refcounts in what could be a sea
2657 of objects that were balanced.
2658 * Added logging to 'make update' command. See update.log
2659 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
2660 do not come from the remote party.
2661 * Added the 'n' option to the SpeechBackground application to tell it to not
2662 answer the channel if it has not already been answered.
2663 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
2664 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
2666 * iLBC source code no longer included (see UPGRADE.txt for details)
2667 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
2668 deadlock is detected, a backtrace of the stack which led to the lock calls
2669 will be output to the CLI.
2670 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
2671 the "core show locks" CLI command will give lock information output as well
2672 as a backtrace of the stack which led to the lock calls.
2673 * users.conf now sports an optional alternateexts property, which permits
2674 allocation of additional extensions which will reach the specified user.
2675 * A new option for the configure script, --enable-internal-poll, has been added
2676 for use with systems which may have a buggy implementation of the poll system
2677 call. If you notice odd behavior such as the CLI being unresponsive on remote
2678 consoles, you may want to try using this option. This option is enabled by default
2679 on Darwin systems since it is known that the Darwin poll() implementation has
2683 --------------------
2684 * In addition to timing from DAHDI, there is a new timing module called
2685 res_timing_timerfd. In order to use this, you must be running Linux with
2686 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
2687 script will be able to tell if you have the requirements. From menuselect, select
2688 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob