1 ==============================================================================
3 === This file documents the new and/or enhanced functionality added in
4 === the Asterisk versions listed below. This file does NOT include
5 === changes in behavior that would not be backwards compatible with
6 === previous versions; for that information see the UPGRADE.txt file
7 === and the other UPGRADE files for older releases.
9 ==============================================================================
10 ------------------------------------------------------------------------------
11 --- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
12 ------------------------------------------------------------------------------
15 AMI (Asterisk Manager Interface)
17 * The SIPshowpeer action will now include a 'SubscribeContext' field for a peer
18 in its response if the peer has a subscribe context set.
20 * The SIPqualifypeer action now acknowledges the request once it has established
21 that the request is against a known peer. It also issues a new event,
22 'SIPqualifypeerdone', once the qualify action has been completed.
24 * The PlayDTMF action now supports an optional 'Duration' parameter. This
25 specifies the duration of the digit to be played, in milliseconds.
27 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
28 updates when changes occur instead of requiring the use of pollmailboxes.
30 * CLI Command 'Manager Show Commands' no longer truncates command names longer
31 than 15 characters and no longer shows authorization requirement for commands.
32 'Manager Show Command' now displays the privileges needed for using a given
33 manager command instead.
35 * Added new action "ControlPlayback". The ControlPlayback action allows an AMI
36 client to manipulate audio currently being played back on a channel. The
37 supported operations depend on the application being used to send audio to
38 the channel. When the audio playback was initiated using the ControlPlayback
39 application or CONTROL STREAM FILE AGI command, the audio can be paused,
40 stopped, restarted, reversed, or skipped forward. When initiated by other
41 mechanisms (such as the Playback application), the audio can be stopped,
42 reversed, or skipped forward.
44 * Channel related events now contain a snapshot of channel state, adding new
45 fields to many of these events.
47 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
48 in a future release. Please use the common 'Exten' field instead.
50 * The AMI event 'UserEvent' from app_userevent now contains the channel state
51 fields. The channel state fields will come before the body fields.
53 * The deprecated use of | (pipe) as a separator in the channelvars setting in
54 manager.conf has been removed.
61 * Added general support for busy detection.
63 * Added ECAM command support for Sony Ericsson phones.
67 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
68 using the 'supportpath' setting, either on a global basis or on a peer basis.
69 This setting enables Asterisk to route outgoing out-of-dialog requests via a
70 set of proxies by using a pre-loaded route-set defined by the Path headers in
71 the REGISTER request. See Realtime updates for more configuration information.
75 * The BRIDGE_FEATURES channel variable would previously only set features for
76 the calling party and would set this feature regardless of whether the
77 feature was in caps or in lowercase. Use of a caps feature for a letter
78 will now apply the feature to the calling party while use of a lowercase
79 letter will apply that feature to the called party.
81 * Add support for automixmonitor to the BRIDGE_FEATURES channel variable.
83 * PARKINGSLOT and PARKEDLOT channel variables will now be set for a parked
84 channel even when comebactoorigin=yes
88 * When performing queue pause/unpause on an interface without specifying an
89 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
90 least one member of any queue exists for that interface.
92 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
93 for realtime queue log entries.
97 * Added the 'n' option to MeetMe to prevent application of the DENOISE function
98 to a channel joining a conference. Some channel drivers that vary the number
99 of audio samples in a voice frame will experience significant quality problems
100 if a denoiser is attached to the channel; this option gives them the ability
101 to remove the denoiser without having to unload func_speex.
105 * Add queue available hint. exten => 8501,hint,Queue:markq_avail
106 Note: the suffix '_avail' after the queuename.
107 Reports 'InUse' for no logged in agents or no free agents.
108 Reports 'Idle' when an agent is free.
112 * Redirecting reasons can now be set to arbitrary strings. This means
113 that the REDIRECTING dialplan function can be used to set the redirecting
114 reason to any string. It also allows for custom strings to be read as the
115 redirecting reason from SIP Diversion headers.
119 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
120 will store the path information for that peer when it registers. Realtime
121 tables can also use the 'supportpath' field to enable Path header support.
123 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
124 objectIdentifier. This maps to the supportpath option in sip.conf.
128 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
129 them, an Asterisk-specific version of pjproject needs to be installed.
130 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
134 * Device state for XMPP buddies is now available using the following format:
135 XMPP/<client name>/<buddy address>
136 If any resource is available the device state is considered to be not in use.
137 If no resources exist or all are unavailable the device state is considered
140 ------------------------------------------------------------------------------
141 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
142 ------------------------------------------------------------------------------
146 * The Asterisk build system will now build and install a shared library
147 (libasteriskssl.so) used to wrap various initialization and shutdown functions
148 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
149 that Asterisk can ensure that these functions do *not* get called by any
150 modules that are loaded into Asterisk, since they should only be called once
151 in any single process. If desired, this feature can be disabled by supplying
152 the "--disable-asteriskssl" option to the configure script.
154 * A new make target, 'full', has been added to the Makefile. This performs
155 the same compilation actions as make all, but will also scan the entirety of
156 each source file for documentation. This option is needed to generate AMI
157 event documentation. Note that your system must have Python in order for
158 this make target to succeed.
160 * The optimization portion of the build system has been reworked to avoid
161 broken builds on certain architectures. All architecture-specific
162 optimization has been removed in favor of using -march=native to allow gcc
163 to detect the environment in which it is running when possible. This can
164 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
166 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
167 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
169 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
170 previously parsed the header file to obtain the version of Asterisk, you
171 will now have to go through Asterisk to get the version information.
179 * Added 'F()' option. Similar to the dial option, this can be supplied with
180 arguments indicating where the callee should go after the caller is hung up,
181 or without options specified, the priority after the Queue will be used.
186 * Added menu action admin_toggle_mute_participants. This will mute / unmute
187 all non-admin participants on a conference. The confbridge configuration
188 file also allows for the default sounds played to all conference users when
189 this occurs to be overriden using sound_participants_unmuted and
190 sound_participants_muted.
192 * Added menu action participant_count. This will playback the number of
193 current participants in a conference.
195 * Added announcement configuration option to user profile. If set the sound
196 file will be played to the user, and only the user, upon joining the
199 * Added record_file_append option that defaults to "yes", but if set to no
200 will create a new file between each start/stop recording.
205 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
206 channels respectively before the callee channels are called.
211 * Added support for IPv6.
213 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
214 external process will cause the current playlist to be cleared, including
215 stopping any audio file that is currently playing. This is useful when you
216 want to interrupt audio playback only when specific DTMF is entered by the
222 * A new option, 'I' has been added to app_followme. By setting this option,
223 Asterisk will not update the caller with connected line changes when they
224 occur. This is similar to app_dial and app_queue.
226 * The 'N' option is now ignored if the call is already answered.
228 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
229 and caller channels respectively before the callee channels are called.
231 * The winning FollowMe outgoing call is now put on hold if the caller put it on
237 * MixMonitor hooks now have IDs associated with them which can be used to
238 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
239 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
240 now accepts that ID as an argument.
242 * Added 'm' option, which stores a copy of the recording as a voicemail in the
248 * The connect action in app_mysql now allows you to specify a port number to
249 connect to. This is useful if you run a MySQL server on a non-standard
255 * Increased the default number of allowed destinations from 5 to 12.
260 * The app_page application now no longer depends on DAHDI or app_meetme. It
261 has been re-architected to use app_confbridge internally.
266 * Added queue options autopausebusy and autopauseunavail for automatically
267 pausing a queue member when their device reports busy or congestion.
269 * The 'ignorebusy' option for queue members has been deprecated in favor of
270 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
271 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
272 per interface basis. Individual ringinuse values can now be set in
273 queues.conf via an argument to member definitions. Lastly, the queue
274 'ringinuse' setting now only determines defaults for the per member
275 'ringinuse' setting and does not override per member settings like it does
278 * Added 'F()' option. Similar to the dial option, this can be supplied with
279 arguments indicating where the callee should go after the caller is hung up,
280 or without options specified, the priority after the Queue will be used.
282 * Added new option log_member_name_as_agent, which will cause the membername to
283 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
284 state_interface has been set.
286 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
290 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
291 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
292 changed arguments to SayUnixTime so that every option is truly optional even
293 when using multiple options (so that j option could be used without having to
294 manually specify timezone and format) There are other benefits, e.g., format
295 can now be used without specifying time zone as well.
300 * Addition of the VM_INFO function - see Function changes.
302 * The imapserver, imapport, and imapflags configuration options can now be
303 overriden on a user by user basis.
305 * When voicemail plays a message's envelope with saycid set to yes, when
306 reaching the caller id field it will play a recording of a file with the same
307 base name as the sender's callerid if there is a similarly named file in
308 <astspooldir>/recordings/callerids/
310 * Voicemails now contains a unique message identifier "msg_id", which is stored
311 in the message envelope with the sound files. IMAP backends will now store
312 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
313 backends will store the message identifier in a "msg_id" column. See
314 UPGRADE.txt for more information.
316 * Added VoiceMailPlayMsg application. This application will play a single
317 voicemail message from a mailbox. The result of the application, SUCCESS or
318 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
323 * Hangup handlers can be attached to channels using the CHANNEL() function.
324 Hangup handlers will run when the channel is hung up similar to the h
325 extension. The hangup_handler_push option will push a GoSub compatible
326 location in the dialplan onto the channel's hangup handler stack. The
327 hangup_handler_pop option will remove the last added location, and optionally
328 replace it with a new GoSub compatible location. The hangup_handler_wipe
329 option will remove all locations on the stack, and optionally add a new
332 * The expression parser now recognizes the ABS() absolute value function,
333 which will convert negative floating point values to positive values.
335 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
336 control of faxdetect.
338 * Addition of the VM_INFO function that can be used to retrieve voicemail
339 user information, such as the email address and full name.
340 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
343 * The REDIRECTING function now supports the redirecting original party id
346 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
347 lets you set some of the configuration options from the [general] section
348 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
349 the key sequence used to activate built-in features, such as blindxfer,
350 and automon. See the built-in documentation for details.
352 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
353 instead of simply the uri. This is the format that MessageSend() can use
354 in the from parameter for outgoing SIP messages.
356 * Added the PRESENCE_STATE function. This allows retrieving presence state
357 information from any presence state provider. It also allows setting
358 presence state information from a CustomPresence presence state provider.
359 See AMI/CLI changes for related commands.
361 * Added the AMI_CLIENT function to make manager account attributes available
362 to the dialplan. It currently supports returning the current number of
363 active sessions for a given account.
365 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
366 and the REDIRECTING functions.
374 * Added a manager event "LocalBridge" for local channel call bridges between
375 the two pseudo-channels created.
380 * Added dialtone_detect option for analog ports to disconnect incoming
381 calls when dialtone is detected.
383 * Added option colp_send to send ISDN connected line information. Allowed
384 settings are block, to not send any connected line information; connect, to
385 send connected line information on initial connect; and update, to send
386 information on any update during a call. Default is update.
388 * Add options namedcallgroup and namedpickupgroup to support installations
389 where a higher number of groups (>64) is required.
391 * Added support to use private party ID information with PRI calls.
396 * A new channel driver named chan_motif has been added which provides support for
397 Google Talk and Jingle in a single channel driver. This new channel driver includes
398 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
399 hold, unhold, and ringing notification. It is also compliant with the current Jingle
400 specification, current Google Jingle specification, and the original Google Talk
406 * Added NAT support for RTP. Setting in config is 'nat', which can be set
407 globally and overriden on a peer by peer basis.
409 * Direct media functionality has been added. Options in config are:
410 directmedia (directrtp) and directrtpsetup (earlydirect)
412 * ChannelUpdate events now contain a CallRef header.
417 * Asterisk will no longer substitute CID number for CID name in the display
418 name field if CID number exists without a CID name. This change improves
419 compatibility with certain device features such as Avaya IP500's directory
422 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
423 created using that setting to not be removed during SIP reload.
425 * Added settings recordonfeature and recordofffeature. When receiving an INFO
426 request with a "Record:" header, this will turn the requested feature on/off.
427 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
428 dynamic features must be enabled and configured properly on the requesting
429 channel for this to function properly.
431 * Add support to realtime for the 'callbackextension' option.
433 * When multiple peers exist with the same address, but differing
434 callbackextension options, incoming requests that are matched by address
435 will be matched to the peer with the matching callbackextension if it is
438 * Two new NAT options, auto_force_rport and auto_comedia, have been added
439 which set the force_rport and comedia options automatically if Asterisk
440 detects that an incoming SIP request crossed a NAT after being sent by
443 * The default global nat setting in sip.conf has been changed from force_rport
446 * NAT settings are now a combinable list of options. The equivalent of the
447 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
449 * Adds an option send_diversion which can be disabled to prevent
450 diversion headers from automatically being added to INVITE requests.
452 * Add support for lightweight NAT keepalive. If enabled a blank packet will
453 be sent to the remote host at a given interval to keep the NAT mapping open.
454 This can be enabled using the keepalive configuration option.
456 * Add option 'tonezone' to specify country code for indications. This option
457 can be set both globally and overridden for specific peers.
459 * The SIP Security Events Framework now supports IPv6.
461 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
462 between multiple user agents. When set, for directmedia reinvites,
463 Asterisk will not send an immediate reinvite on an incoming call leg. This
464 option is useful when peered with another SIP user agent that is known to
465 send immediate direct media reinvites upon call establishment.
467 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
470 * Add options subminexpiry and submaxexpiry to set limits of subscription
471 timer independently from registration timer settings. The setting of the
472 registration timer limits still is done by options minexpiry, maxexpiry
473 and defaultexpiry. For backwards compatibility the setting of minexpiry
474 and maxexpiry also is used to configure the subscription timer limits if
475 subminexpiry and submaxexpiry are not set in sip.conf.
477 * Set registration timer limits to default values when reloading sip
478 configuration and values are not set by configuration.
480 * Add options namedcallgroup and namedpickupgroup to support installations
481 where a higher number of groups (>64) is required.
483 * When a MESSAGE request is received, the address the request was received from
484 is now saved in the SIP_RECVADDR variable.
486 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
487 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
488 the ANI2/OLI information is set on the channel, which can be retrieved using
489 the CALLERID function.
491 * Peers can now be configured to support negotiation of ICE candidates using
492 the setting icesupport. See res_rtp_asterisk changes for more information.
494 * Added support for format attribute negotiation. See the Codecs changes for
497 * Extra headers specified with SIPAddHeader are sent with the REFER message
498 when using Transfer application. See refer_addheaders in sip.conf.sample.
500 * Added support to use private party ID information with calls.
502 * Adds an option discard_remote_hold_retrieval that when set stops telling
503 the peer to start music on hold.
508 * Added skinny version 17 protocol support.
513 * Added ability to use multiple lines for a single phone. This allows multiple
514 calls to occur on a single phone, using callwaiting and switching between calls.
516 * Added option 'sharpdial' allowing end dialing by pressing # key
518 * Added option 'interdigit_timer' to control phone dial timeout
520 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
522 * Added global 'debug' option, that enables debug in channel driver
524 * Added ability to translate on-screen menu in multiple languages. Tested on
525 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
526 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
529 * In addition to English added French and Russian languages for on-screen menus
531 * Reworked dialing number input: added dialing by timeout, immediate dial on
532 on dialplan compare, phone number length now not limited by screen size
534 * Added ability to pickup a call using features.conf defined value and
540 * Add options namedcallgroup and namedpickupgroup to support installations
541 where a higher number of groups (>64) is required.
543 * Added support to use private party ID information with calls.
548 * The minimum DTMF duration can now be configured in asterisk.conf
549 as "mindtmfduration". The default value is (as before) set to 80 ms.
550 (previously it was only available in source code)
552 * Named ACLs can now be specified in acl.conf and used in configurations that
553 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
554 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
555 working ACL. In addition, some CLI commands have been added to provide
556 show information and allow for module reloading - see CLI Changes.
558 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
559 items (separated by commas), and items in the rule can be negated by prefixing
560 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
561 longer necessray to control the order that the 'permit' and 'deny' columns are
562 returned from queries.
564 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
565 be used within the dynamic weight attribute when specifying a mapping.
567 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
568 header, instead of putting the user defined event name there. When enabled
569 the UserDefType header is added for user defined events. This feature is
570 enabled with the setting show_user_defined.
572 * Macro has been deprecated in favor of GoSub. For redirecting and connected
573 line purposes use the following variables instead of their macro equivalents:
574 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
575 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
576 cc_callback_macro in channel configurations.
578 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
581 * Call files now support the "early_media" option to connect with an outgoing
582 extension when early media is received.
584 * Added support to use private party ID information with calls.
589 * A new channel variable, AGIEXITONHANGUP, has been added which allows
590 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
591 AGI application would exit immediately after a channel hangup is detected.
593 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
594 are resolved and each address is attempted in turn until one succeeds or
598 AMI (Asterisk Manager Interface)
600 * The originate action now has an option "EarlyMedia" that enables the
601 call to bridge when we get early media in the call. Previously,
602 early media was disregarded always when originating calls using AMI.
604 * Added setvar= option to manager accounts (much like sip.conf)
606 * Originate now generates an error response if the extension given is not found
609 * MixMonitor will now show IDs associated with the mixmonitor upon creating
610 them if the i(variable) option is used. StopMixMonitor will accept
611 MixMonitorID as an option to close specific MixMonitors.
613 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
614 updated to include information about peers configured with
615 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
616 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
617 returned if auto_force_rport is not enabled.
619 * Added SIPpeerstatus manager command which will generate PeerStatus events
620 similar to the existing PeerStatus events found in chan_sip on demand.
622 * Hangup now can take a regular expression as the Channel option. If you want
623 to hangup multiple channels, use /regex/ as the Channel option. Existing
624 behavior to hanging up a single channel is unchanged, but if you pass a regex,
625 the manager will send you a list of channels back that were hung up.
627 * Support for IPv6 addresses has been added.
629 * AMI Events can now be documented in the Asterisk source. Note that AMI event
630 documentation is only generated when Asterisk is compiled using 'make full'.
631 See the CLI section for commands to display AMI event information.
633 * The AMI Hangup event now includes the AccountCode header so you can easily
634 correlate with AMI Newchannel events.
636 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
637 the StateInterface of the queue member.
639 * Added AMI event SessionTimeout in the Call category that is issued when a
640 call is terminated due to either RTP stream inactivity or SIP session timer
643 * CEL events can now contain a user defined header UserDefType. See core
644 changes for more information.
646 * OOH323 ChannelUpdate events now contain a CallRef header.
648 * Added PresenceState command. This command will report the presence state for
649 the given presence provider.
651 * Added Parkinglots command. This will list all parking lots as a series of
652 AMI Parkinglot events.
654 * Added MessageSend command. This behaves in the same manner as the
655 MessageSend application, and is a technolgoy agnostic mechanism to send out
656 of call text messages.
658 * Added "message" class authorization. This grants an account permission to
659 send out of call messages. Write-only.
664 * The "dialplan add include" command has been modified to create context a context
665 if one does not already exist. For instance, "dialplan add include foo into bar"
666 will create context "bar" if it does not already exist.
668 * A "dialplan remove context" command has been added to remove a context from
671 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
672 filenames of all running mixmonitors on a channel.
674 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
675 numeric instead of 0, 1, or 2.
677 * "stun show status" will show a table describing how the STUN client is
680 * "acl show [named acl]" will show information regarding a Named ACL. The
681 acl module can be reloaded with "reload acl".
683 * Added CLI command to display AMI event information - "manager show events",
684 which shows a list of all known and documented AMI events, and "manager show
685 event [event name]", which shows detail information about a specific AMI
688 * The result of the CLI command "queue show" now includes the state interface
689 information of the queue member.
691 * The command "core set verbose" will now set a separate level of logging for
692 each remote console without affecting any other console.
694 * Added command "cdr show pgsql status" to check connection status
696 * "sip show channel" will now display the complete route set.
698 * Added "presencestate list" command. This command will list all custom
699 presence states that have been set by using the PRESENCE_STATE dialplan
702 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
703 command. This changes a custom presence to a new state.
708 * Codec lists may now be modified by the '!' character, to allow succinct
709 specification of a list of codecs allowed and disallowed, without the
710 requirement to use two different keywords. For example, to specify all
711 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
713 * Add support for parsing SDP attributes, generating SDP attributes, and
714 passing it through. This support includes codecs such as H.263, H.264, SILK,
715 and CELT. You are able to set up a call and have attribute information pass.
716 This should help considerably with video calls.
718 * The iLBC codec can now use a system-provided iLBC library if one is installed,
719 just like the GSM codec.
723 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
724 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
728 * Asterisk version and build information is now logged at the beginning of a
731 * Threads belonging to a particular call are now linked with callids which get
732 added to any log messages produced by those threads. Log messages can now be
733 easily identified as involved with a certain call by looking at their call id.
734 Call ids may also be attached to log messages for just about any case where
735 it can be determined to be related to a particular call.
737 * Each logging destination and console now have an independent notion of the
738 current verbosity level. Logger.conf now allows an optional argument to
739 the 'verbose' specifier, indicating the level of verbosity sent to that
740 particular logging destination. Additionally, remote consoles now each
741 have their own verbosity level. The command 'core set verbose' will now set
742 a separate level for each remote console without affecting any other
748 * Added 'announcement' option which will play at the start of MOH and between
749 songs in modes of MOH that can detect transitions between songs (eg.
755 * New per parking lot options: comebackcontext and comebackdialtime. See
756 configs/features.conf.sample for more details.
758 * Channel variable PARKER is now set when comebacktoorigin is disabled in
761 * Channel variable PARKEDCALL is now set with the name of the parking lot
762 when a timeout occurs.
768 CDR Postgresql Driver
770 * Added command "cdr show pgsql status" to check connection status
773 CDR Adaptive ODBC Driver
775 * Added schema option for databases that support specifying a schema.
783 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
784 CALENDAR_WRITE has completed successfully.
789 * A new option, 'probation' has been added to rtp.conf
790 RTP in strictrtp mode can now require more than 1 packet to exit learning
791 mode with a new source (and by default requires 4). The probation option
792 allows the user to change the required number of packets in sequence to any
793 desired value. Use a value of 1 to essentially restore the old behavior.
794 Also, with strictrtp on, Asterisk will now drop all packets until learning
795 mode has successfully exited. These changes are based on how pjmedia handles
796 media sources and source changes.
798 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
799 enabled or disabled using the icesupport setting. A variety of other
800 settings have been introduced to configure STUN/TURN connections.
805 * A new module, res_corosync, has been introduced. This module uses the
806 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
807 of Asterisk servers to both Message Waiting Indication (MWI) and/or
808 Device State (presence) information. This module is very similar to, and
809 is a replacement for the res_ais module that was in previous releases of
815 * This module adds a cleaned up, drop-in replacement for res_jabber called
816 res_xmpp. This provides the same externally facing functionality but is
817 implemented differently internally. res_jabber has been deprecated in favor
818 of res_xmpp; please see the UPGRADE.txt file for more information.
823 * The safe_asterisk script has been updated to allow several of its parameters
824 to be set from environment variables. This also enables a custom run
825 directory of Asterisk to be specified, instead of defaulting to /tmp.
827 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
828 its value to determine the directory to assume is the top-level directory of
829 the source tree. If the variable is not set, it defaults to the current
830 behavior and uses the current working directory.
832 ------------------------------------------------------------------------------
833 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
834 ------------------------------------------------------------------------------
838 * Asterisk now has protocol independent support for processing text messages
839 outside of a call. Messages are routed through the Asterisk dialplan.
840 SIP MESSAGE and XMPP are currently supported. There are options in
841 jabber.conf and sip.conf to allow enabling these features.
842 -> jabber.conf: see the "sendtodialplan" and "context" options.
843 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
844 and "outofcall_message_context" options.
845 The MESSAGE() dialplan function and MessageSend() application have been
846 added to go along with this functionality. More detailed usage information
847 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
848 * If real-time text support (T.140) is negotiated, it will be preferred for
849 sending text via the SendText application. For example, via SIP, messages
850 that were once sent via the SIP MESSAGE request would be sent via RTP if
851 T.140 text is negotiated for a call.
855 * parkedmusicclass can now be set for non-default parking lots.
857 Asterisk Manager Interface
858 --------------------------
859 * PeerStatus now includes Address and Port.
860 * Added Hold events for when the remote party puts the call on and off hold
861 for chan_dahdi ISDN channels.
862 * Added new action MeetmeListRooms to list active conferences (shows same
863 data as "meetme list" at the CLI).
864 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
865 Description field that is set by 'description' in the channel configuration
867 * Added Uniqueid header to UserEvent.
868 * Added new action FilterAdd to control event filters for the current session.
869 This requires the system permission and uses the same filter syntax as
870 filters that can be defined in manager.conf
871 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
872 versions had some instances of the event converted, but others were left
873 as-is. All Unlink events should now be converted to Bridge events. The AMI
874 protocol version number was incremented to 1.2 as a result of this change.
877 --------------------------
878 * The HTTP Server can bind to IPv6 addresses.
881 --------------------------
882 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
883 with busydetect. usage example: busypattern=200,200,200,600
886 --------------------------
887 * New 'gtalk show settings' command showing the current settings loaded from
889 * The 'logger reload' command now supports an optional argument, specifying an
890 alternate configuration file to use.
891 * 'dialplan add extension' command will now automatically create a context if
892 the specified context does not exist with a message indicated it did so.
893 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
894 Description field which can be populated with 'description' in the channel
895 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
898 --------------------------
899 * The filter option in cdr_adaptive_odbc now supports negating the argument,
900 thus allowing records which do NOT match the specified filter.
901 * Added ability to log CONGESTION calls to CDR
904 --------------------------
905 * Ability to define custom SILK formats in codecs.conf.
906 * Addition of speex32 audio format with translation.
907 * CELT codec pass-through support and ability to define
908 custom CELT formats in codecs.conf.
909 * Ability to read raw signed linear files with sample rates
910 ranging from 8khz - 192khz. The new file extensions introduced
911 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
912 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
913 Skinny, H.323, etc) can still only support the following codecs:
914 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
915 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
916 Video: h261, h263, h263p, h264, mpeg4
921 --------------------------
922 * New highly optimized and customizable ConfBridge application capable of
923 mixing audio at sample rates ranging from 8khz-96khz.
924 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
925 and bridge profiles on a channel.
926 * CONFBRIDGE_INFO dialplan function capable of retrieving information
927 about a conference such as locked status and number of parties, admins,
929 * Addition of video_mode option in confbridge.conf for adding video support
930 into a bridge profile.
931 * Addition of the follow_talker video_mode in confbridge.conf. This video
932 mode dynamically switches the video feed to always display the loudest talker
933 supplying video in the conference.
937 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
938 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
939 variables from asterisk.conf.
943 * Addition of the JITTERBUFFER dialplan function. This function allows
944 for jitterbuffering to occur on the read side of a channel. By using
945 this function conference applications such as ConfBridge and MeetMe can
946 have the rx streams jitterbuffered before conference mixing occurs.
947 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
949 * Added STRREPLACE function. This function let's the user search a variable
950 for a given string to replace with another string as many times as the
951 user specifies or just throughout the whole string.
952 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
953 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
954 * Added extensions to chan_ooh323 in function CHANNEL()
956 libpri channel driver (chan_dahdi) DAHDI changes
957 --------------------------
958 * Added moh_signaling option to specify what to do when the channel's bridged
959 peer puts the ISDN channel on hold.
960 * Added display_send and display_receive options to control how the display ie
961 is handled. To send display text from the dialplan use the SendText()
962 application when the option is enabled.
963 * Added mcid_send option to allow sending a MCID request on a span.
966 --------------------------
967 * Added setvar option to calendar.conf to allow setting channel variables on
968 notification channels.
969 * Added "calendar show types" CLI command to list registered calendar
973 --------------------------
974 * Added two new options, r and t with file name arguments to record
975 single direction (unmixed) audio recording separate from the bidirectional
976 (mixed) recording. The mixed file name argument is optional now as long
977 as at least one recording option is used.
980 --------------------------
981 * Added a new option, l, which will disable local call optimization for
982 channels involved with the FollowMe thread. Use this option to improve
983 compatability for a FollowMe call with certain dialplan apps, options, and
987 --------------------------
988 * Added option "k" that will automatically close the conference when there's
989 only one person left when a user exits the conference.
992 --------------------------
993 * cel_pgsql now supports the 'extra' column for data added using the
994 CELGenUserEvent() application.
997 --------------------------
998 * Support for defining hints has been added to pbx_lua. See the 'hints' table
999 in the sample extensions.lua file for syntax details.
1000 * Applications that perform jumps in the dialplan such as Goto will now
1001 execute properly. When pbx_lua detects that the context, extension, or
1002 priority we are executing on has changed it will immediately return control
1003 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
1004 the priority after the currently executing priority.
1005 * An autoservice is now started by default for pbx_lua channels. It can be
1006 stopped and restarted using the autoservice_stop() and autoservice_start()
1010 --------------------------
1011 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
1012 into a FAXStatus event with an 'Operation' header that will be either
1013 'send', 'receive', and 'gateway'.
1014 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
1015 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
1016 feature will handle converting a fax call between an audio T.30 fax terminal
1017 and an IFP T.38 fax terminal.
1021 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
1022 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
1023 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
1027 * Added general option negative_penalty_invalid default off. when set
1028 members are seen as invalid/logged out when there penalty is negative.
1029 for realtime members when set remove from queue will set penalty to -1.
1030 * Added queue option autopausedelay when autopause is enabled it will be
1031 delayed for this number of seconds since last successful call if there
1032 was no prior call the agent will be autopaused immediately.
1033 * Added member option ignorebusy this when set and ringinuse is not
1034 will allow per member control of multiple calls as ringinuse does for
1039 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
1041 * Added 'k' option to MeetMe to automatically kill the conference when there's only
1042 one participant left (much like a normal call bridge)
1043 * Added extra argument to Originate to set timeout.
1047 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
1048 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
1049 utility in the UTILS section of menuselect. If an existing astdb is found and no
1050 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
1051 convert an existing astdb to the SQLite3 version automatically at runtime.
1055 * Modules marked as deprecated are no longer marked as building by default. Enabling
1056 these modules is still available via menuselect.
1060 * authdebug is now disabled by default. To enable this functionaility again
1061 set authdebug = yes in iax.conf.
1065 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
1066 releases it was disabled.
1070 * The PBX core previously made a call with a non-existing extension test for
1071 extension s@default and jump there if the extension existed.
1072 This was a bad default behaviour and violated the principle of least surprise.
1073 It has therefore been changed in this release. It may affect some
1074 applications and configurations that rely on this behaviour. Most channel
1075 drivers have avoided this for many releases by testing whether the extension
1076 called exists before starting the PBX and generating a local error.
1077 This behaviour still exists and works as before.
1079 Extension "s" is used when no extension is given in a channel driver,
1080 like immediate answer in DAHDI or calling to a domain with no user part
1083 ------------------------------------------------------------------------------
1084 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
1085 ------------------------------------------------------------------------------
1089 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
1090 now defaults to force_rport. It is very important that phones requiring nat=no be
1091 specifically set as such instead of relying on the default setting. If at all
1092 possible, all devices should have nat settings configured in the general section as
1093 opposed to configuring nat per-device.
1094 * Added preferred_codec_only option in sip.conf. This feature limits the joint
1095 codecs sent in response to an INVITE to the single most preferred codec.
1096 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
1097 to be used for the outgoing call. It must be one of the codecs configured
1099 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
1100 to be used for holding a private key. If tlsprivatekey is not specified,
1101 tlscertfile is searched for both public and private key.
1102 * Added tlsclientmethod option to sip.conf. This allows the protocol for
1103 outbound client connections to be specified.
1104 * The sendrpid parameter has been expanded to include the options
1105 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
1106 header to be sent (equivalent to setting sendrpid=yes) and setting
1107 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
1108 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
1109 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
1110 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
1111 will accept the SDP even if the SDP version number is not properly incremented,
1112 but will generate a warning in the log indicating that the SIP peer that sent
1113 the SDP should have the 'ignoresdpversion' option set.
1114 * The 'nat' option has now been been changed to have yes, no, force_rport, and
1115 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
1116 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
1117 remote side requests it and disables symmetric RTP support. Setting it to
1118 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
1119 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
1120 and enables symmetric RTP support.
1121 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
1122 response. This permits the master channel to know how each channel dialled
1123 in a multi-channel setup resolved in an individual way. This carries a
1124 performance penalty and can be disabled in sip.conf using the
1125 'storesipcause' option.
1126 * Added 'externtcpport' and 'externtlsport' options to allow custom port
1127 configuration for the externip and externhost options when tcp or tls is used.
1128 * Added support for message body (stored in content variable) to SIP NOTIFY message
1129 accessible via AMI and CLI.
1130 * Added 'media_address' configuration option which can be used to explicitly specify
1131 the IP address to use in the SDP for media (audio, video, and text) streams.
1132 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
1133 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
1135 * Added 'use_q850_reason' configuration option for generating and parsing
1136 if available Reason: Q.850;cause=<cause code> header. It is implemented
1137 in some gateways for better passing PRI/SS7 cause codes via SIP.
1138 * When dialing SIP peers, a new component may be added to the end of the dialstring
1139 to indicate that a specific remote IP address or host should be used when dialing
1140 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
1141 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
1142 ability to selectively force bridged channels to also be encrypted is also
1143 implemented. Branching in the dialplan can be done based on whether or not
1144 a channel has secure media and/or signaling.
1145 * Added directmediapermit/directmediadeny to limit which peers can send direct media
1147 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
1148 Charge messages to snom phones.
1149 * Added support for G.719 media streams.
1150 * Added support for 16khz signed linear media streams.
1151 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
1152 RTP has been outfitted with the same abilities.
1153 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
1154 available in device configurations as well as in the dial plan.
1155 * Addition of the 'subscribe_network_change' option for turning on and off
1156 res_stun_monitor module support in chan_sip.
1157 * Addition of the 'auth_options_requests' option for turning on and off
1158 authentication for OPTIONS requests in chan_sip.
1162 * Add #tryinclude statement for config files. This provides the same
1163 functionality as the #include statement however an asterisk module will
1164 still load if the filename does not exist. Using the #include statement
1165 Asterisk will not allow the module to load.
1169 * Added rtsavesysname option into iax.conf to allow the systname to be saved
1170 on realtime updates.
1171 * Added the ability for chan_iax2 to inform the dialplan whether or not
1172 encryption is being used. This interoperates with the SIP SRTP implementation
1173 so that a secure SIP call can be bridged to a secure IAX call when the
1174 dialplan requires bridged channels to be "secure".
1175 * Addition of the 'subscribe_network_change' option for turning on and off
1176 res_stun_monitor module support in chan_iax.
1181 * Added ability to preset channel variables on indicated lines with the setvar
1182 configuration option. Also, clearvars=all resets the list of variables back
1184 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
1185 See configs/res_pktccops.conf for more information.
1187 XMPP Google Talk/Jingle changes
1188 -------------------------------
1189 * Added the externip option to gtalk.conf.
1190 * Added the stunaddr option to gtalk.conf which allows for the automatic
1191 retrieval of the external ip from a stun server.
1195 * Added 'p' option to PickupChan() to allow for picking up channel by the first
1196 match to a partial channel name.
1197 * Added .m3u support for Mp3Player application.
1198 * Added progress option to the app_dial D() option. When progress DTMF is
1199 present, those values are sent immediately upon receiving a PROGRESS message
1200 regardless if the call has been answered or not.
1201 * Added functionality to the app_dial F() option to continue with execution
1202 at the current location when no parameters are provided.
1203 * Added the 'a' option to app_dial to answer the calling channel before any
1204 announcements or macros are executed.
1205 * Modified app_dial to set answertime when the called channel answers even if
1206 the called channel hangs up during playback of an announcement.
1207 * Modified app_dial 'r' option to support an additional parameter to play an
1208 indication tone from indications.conf
1209 * Added c() option to app_chanspy. This option allows custom DTMF to be set
1210 to cycle through the next available channel. By default this is still '*'.
1211 * Added x() option to app_chanspy. This option allows DTMF to be set to
1212 exit the application.
1213 * The Voicemail application has been improved to automatically ignore messages
1214 that only contain silence.
1215 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
1216 associated mailbox(es) to be greetings-only.
1217 * The ChanSpy application now has the 'S' option, which makes the application
1218 automatically exit once it hits a point where no more channels are available
1220 * The ChanSpy application also now has the 'E' option, which spies on a single
1221 channel and exits when that channel hangs up.
1222 * The MeetMe application now turns on the DENOISE() function by default, for
1223 each participant. In our tests, this has significantly decreased background
1224 noise (especially noisy data centers).
1225 * Voicemail now permits storage of secrets in a separate file, located in the
1226 spool directory of each individual user. The control for this is located in
1227 the "passwordlocation" option in voicemail.conf. Please see the sample
1228 configuration for more information.
1229 * The ChanIsAvail application now exposes the returned cause code using a separate
1230 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
1231 * Added 'd' option to app_followme. This option disables the "Please hold"
1233 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
1234 received will terminate recording.
1235 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
1236 Previously the folder could only be set per context, but has now been extended
1237 using the imapfolder option.
1238 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
1239 * Voicemail now allows the pager date format to be specified separately from the
1241 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
1242 to allow joining, leaving, and sending text to group chats.
1243 * MeetMe has a new option 'G' to play an announcement before joining a conference.
1244 * Page has a new option 'A(x)' which will playback an announcement simultaneously
1245 to all paged phones (and optionally excluding the caller's one using the new
1246 option 'n') before the call is bridged.
1247 * The 'f' option to Dial has been augmented to take an optional argument. If no
1248 argument is provided, the 'f' option works as it always has. If an argument is
1249 provided, then the connected party information of all outgoing channels created
1250 during the Dial will be set to the argument passed to the 'f' option.
1251 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
1253 * The OSP lookup application adds in/outbound network ID, optional security,
1254 number portability, QoS reporting, destination IP port, custom info and service
1256 * Added new application VMSayName that will play the recorded name of the voicemail
1257 user if it exists, otherwise will play the mailbox number.
1258 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
1259 retrieve state for a particular bridge, where <name> is the conference name
1260 * app_directory now allows exiting at any time using the operator or pound key.
1261 * Voicemail now supports setting a locale per-mailbox.
1262 * Two new applications are provided for declining counting phrases in multiple
1263 languages. See the application notes for SayCountedNoun and SayCountedAdj for
1265 * Voicemail now runs the externnotify script when pollmailboxes is activated and
1267 * Voicemail now includes rdnis within msgXXXX.txt file.
1268 * ExternalIVR now supports IPv6 addresses.
1269 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
1270 at https://wiki.asterisk.org/wiki/x/oQBB
1271 * ParkedCall and Park can now specify the parking lot to use.
1275 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
1276 over SRV records associated with a specific service. From the CLI, type
1277 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
1278 details on how these may be used.
1279 * PITCH_SHIFT dialplan function added. This function can be used to modify the
1280 pitch of a channel's tx and rx audio streams.
1281 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
1282 setting various connected line and redirecting party information.
1283 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
1284 support ISDN subaddressing.
1285 * The CHANNEL() function now supports the "name" and "checkhangup" options.
1286 * For DAHDI channels, the CHANNEL() dialplan function now allows
1287 the dialplan to request changes in the configuration of the active
1288 echo canceller on the channel (if any), for the current call only.
1291 exten => s,n,Set(CHANNEL(echocan_mode)=off)
1293 The possible values are:
1295 on - normal mode (the echo canceller is actually reinitialized)
1297 fax - FAX/data mode (NLP disabled if possible, otherwise completely
1299 voice - voice mode (returns from FAX mode, reverting the changes that
1300 were made when FAX mode was requested)
1301 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
1302 and setting variables on the channel which created the current channel.
1303 Administrators should take care to avoid naming conflicts, when multiple
1304 channels are dialled at once, especially when used with the Local channel
1305 construct (which all could set variables on the master channel). Usage
1306 of the HASH() dialplan function, with the key set to the name of the slave
1307 channel, is one approach that will avoid conflicts.
1308 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
1310 * func_odbc now allows multiple row results to be retrieved without using
1311 mode=multirow. If rowlimit is set, then additional rows may be retrieved
1312 from the same query by using the name of the function which retrieved the
1313 first row as an argument to ODBC_FETCH().
1314 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
1315 dialplan. This function returns the content of the received message.
1316 * Added REPLACE, which searches a given variable name for a set of characters,
1317 then either replaces them with a single character or deletes them.
1318 * Added PASSTHRU, which literally passes the same argument back as its return
1319 value. The intent is to be able to use a literal string argument to
1320 functions that currently require a variable name as an argument.
1321 * HASH-associated variables now can be inherited across channel creation, by
1322 prefixing the name of the hash at assignment with the appropriate number of
1323 underscores, just like variables.
1324 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
1325 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
1326 whether or not channels that are bridged to the current channel will be
1327 required to have secure signaling and/or media.
1328 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
1329 the current channel has secure signaling and/or media.
1330 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
1331 "no_media_path" option.
1332 Returns "0" if there is a B channel associated with the call.
1333 Returns "1" if no B channel is associated with the call. The call is either
1334 on hold or is a call waiting call.
1335 * Added option to dialplan function CDR(), the 'f' option
1336 allows for high resolution times for billsec and duration fields.
1337 * FILE() now supports line-mode and writing.
1338 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
1339 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
1343 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
1344 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
1345 and is set when a dynamic feature is triggered.
1346 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
1347 to dynamically create a new parking lot matching the value this varible is
1349 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
1350 features.conf that should be the base for dynamic parkinglots.
1351 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
1352 parkinglot should have.
1353 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
1354 parkinglot should have.
1355 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
1360 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
1361 timeout has expired.
1362 * Added 'R' option to app_queue. This option stops moh and indicates ringing
1363 to the caller when an Agent's phone is ringing. This can be used to indicate
1364 to the caller that their call is about to be picked up, which is nice when
1365 one has been on hold for an extened period of time.
1366 * A new config option, penaltymemberslimit, has been added to queues.conf.
1367 When set this option will disregard penalty settings when a queue has too
1369 * A new option, 'I' has been added to both app_queue and app_dial.
1370 By setting this option, Asterisk will not update the caller with
1371 connected line changes or redirecting party changes when they occur.
1372 * A 'relative-periodic-announce' option has been added to queues.conf. When
1373 enabled, this option will cause periodic announce times to be calculated
1374 from the end of announcements rather than from the beginning.
1375 * The autopause option in queues.conf can be passed a new value, "all." The
1376 result is that if a member becomes auto-paused, he will be paused in all
1377 queues for which he is a member, not just the queue that failed to reach
1379 * Added dialplan function QUEUE_EXISTS to check if a queue exists
1380 * The queue logger now allows events to optionally propagate to a file,
1381 even when realtime logging is turned on. Additionally, realtime logging
1382 supports sending the event arguments to 5 individual fields, although it
1383 will fallback to the previous data definition, if the new table layout is
1386 mISDN channel driver (chan_misdn) changes
1387 ----------------------------------------
1388 * Added display_connected parameter to misdn.conf to put a display string
1389 in the CONNECT message containing the connected name and/or number if
1390 the presentation setting permits it.
1391 * Added display_setup parameter to misdn.conf to put a display string
1392 in the SETUP message containing the caller name and/or number if the
1393 presentation setting permits it.
1394 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
1395 indicate the dialplan settings are to be obtained from the asterisk
1397 * Made misdn.conf parameter callerid accept the "name" <number> format
1398 used by the rest of the system.
1399 * Made use the nationalprefix and internationalprefix misdn.conf
1400 parameters to prefix any received number from the ISDN link if that
1401 number has the corresponding Type-Of-Number. NOTE: This includes
1402 comparing the incoming call's dialed number against the MSN list.
1403 * Added the following new parameters: unknownprefix, netspecificprefix,
1404 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
1405 received number from the ISDN link if that number has the corresponding
1407 * Added new dialplan application misdn_command which permits controlling
1408 the CCBS/CCNR functionality.
1409 * Added new dialplan function mISDN_CC which permits retrieval of various
1410 values from an active call completion record.
1411 * For PTP, you should manually send the COLR of the redirected-to party
1412 for an incomming redirected call if the incoming call could experience
1413 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
1414 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
1415 if the REDIRECTING(from-num) is not empty.
1416 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
1417 option on all of the REDIRECTING statements before dialing the
1418 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
1419 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
1420 redirecting-to presentation (COLR) when it becomes available.
1421 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
1424 thirdparty mISDN enhancements
1425 -----------------------------
1426 mISDN has been modified by Digium, Inc. to greatly expand facility message
1428 * Enhanced COLP support for call diversion and transfer.
1429 * CCBS/CCNR support.
1431 The latest modified mISDN v1.1.x based version is available at:
1432 http://svn.digium.com/svn/thirdparty/mISDN/trunk
1433 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
1435 Tagged versions of the modified mISDN code are available under:
1436 http://svn.digium.com/svn/thirdparty/mISDN/tags
1437 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
1439 libpri channel driver (chan_dahdi) DAHDI changes
1440 -------------------------------------------
1441 * The channel variable PRIREDIRECTREASON is now just a status variable
1442 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
1443 to read and alter the reason.
1444 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
1445 redirected-to party for an incomming redirected call if the incoming call
1446 could experience further redirects. Just set the
1447 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
1448 to the COLR. A call has been redirected if the REDIRECTING(count) is not
1450 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
1451 use the inhibit(i) option on all of the REDIRECTING statements before
1452 dialing the redirected-to party. You still have to set the
1453 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
1454 will update the redirecting-to presentation (COLR) when it becomes available.
1455 * Added the ability to ignore calls that are not in a Multiple Subscriber
1456 Number (MSN) list for PTMP CPE interfaces.
1457 * Added dynamic range compression support for dahdi channels. It is
1458 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
1459 * Added support for ISDN calling and called subaddress with partial support
1460 for connected line subaddress.
1461 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
1462 * Added handling of received HOLD/RETRIEVE messages and the optional ability
1463 to transfer a held call on disconnect similar to an analog phone.
1464 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
1465 Will reroute/deflect an outgoing call when receive the message.
1466 Can use the DAHDISendCallreroutingFacility to send the message for the
1468 * Added standard location to add options to chan_dahdi dialing:
1469 Dial(DAHDI/g1[/extension[/options]])
1472 R Reverse charging indication
1473 * Added Reverse Charging Indication (Collect calls) send/receive option.
1474 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
1475 Dial(DAHDI/g1/extension/R)
1476 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
1477 (requires latest LibPRI)
1478 * Added ability to send/receive keypad digits in the SETUP message.
1479 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
1480 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
1481 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
1482 (requires latest LibPRI)
1483 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
1484 to eliminate tromboned calls. A tromboned call goes out an interface and comes
1485 back into the same interface. Tromboned calls happen because of call routing,
1486 call deflection, call forwarding, and call transfer.
1487 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
1488 * Added the ability to support call waiting calls. (The SETUP has no B channel
1490 * Added Malicious Call ID (MCID) event to the AMI call event class.
1491 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
1493 Asterisk Manager Interface
1494 --------------------------
1495 * The Hangup action now accepts a Cause header which may be used to
1496 set the channel's hangup cause.
1497 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
1498 to specify a separate .pem file to hold a private key. By default sslcert
1499 is used to hold both the public and private key.
1500 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
1501 for options containing the 'tls' prefix. For example, 'sslenable' is now
1502 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
1503 across all .conf files. All affected sample.conf files have been modified to
1504 reflect this change. Previous options such as 'sslenable' still work,
1505 but options with the 'tls' prefix are preferred.
1506 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
1507 in a channel. (res_mutestream.so)
1508 * The configuration file manager.conf now supports a channelvars option, which
1509 specifies a list of channel variables to include in each channel-oriented
1511 * The redirect command now has new parameters ExtraContext, ExtraExtension,
1512 and ExtraPriority to allow redirecting the second channel to a different
1513 location than the first.
1514 * Added new event "JabberStatus" in the Jabber module to monitor buddies
1516 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
1517 in a MixMonitor recording.
1518 * The 'iax2 show peers' output is now similar to the expected output of
1520 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
1522 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
1523 AOC-E messages on a channel.
1524 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
1525 conform more closely to similar events.
1526 * Added a new eventfilter option per user to allow whitelisting and blacklisting
1528 * Added optional parkinglot variable for park command.
1529 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
1530 if CallerIDNum and CallerIDName headers are also present.
1532 Channel Event Logging
1533 ---------------------
1534 * A new interface, CEL, is introduced here. CEL logs single events, much like
1535 the AMI, but it differs from the AMI in that it logs to db backends much
1536 like CDR does; is based on the event subsystem introduced by Russell, and
1537 can share in all its benefits; allows multiple backends to operate like CDR;
1538 is specialized to event data that would be of concern to billing sytems,
1539 like CDR. Backends for logging and accounting calls have been produced,
1540 but a new CDR backend is still in development.
1544 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
1545 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
1546 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
1547 * Multiple files and formats can now be specified in cdr_custom.conf.
1548 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
1549 See configs/cdr_syslog.conf.sample for more information.
1550 * A 'sequence' field has been added to CDRs which can be combined with
1551 linkedid or uniqueid to uniquely identify a CDR.
1552 * Handling of billsec and duration field has changed. If your table definition
1553 specifies those fields as float,double or similar they will now be logged with
1554 microsecond accuracy instead of a whole integer.
1556 Calendaring for Asterisk
1557 ------------------------
1558 * A new set of modules were added supporing calendar integration with Asterisk.
1559 Dialplan functions for reading from and writing to calendars are included,
1560 as well as the ability to execute dialplan logic upon calendar event notifications.
1561 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
1562 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
1563 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
1564 2003 support does not support forms-based authentication).
1566 Call Completion Supplementary Services for Asterisk
1567 ---------------------------------------------------
1568 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
1569 DAHDI/ISDN supports call completion for the following switch types:
1570 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
1571 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
1573 Multicast RTP Support
1574 ---------------------
1575 * A new RTP engine and channel driver have been added which supports Multicast RTP.
1576 The channel driver can be used with the Page application to perform multicast RTP
1577 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
1578 Type can be either basic or linksys.
1579 Destination is the IP address and port for the RTP packets.
1580 Control address is specific to the linksys type and is used for sending the control
1581 packets unique to them.
1583 Security Events Framework
1584 -------------------------
1585 * Asterisk has a new C API for reporting security events. The module res_security_log
1586 sends these events to the "security" logger level. Currently, AMI is the only
1587 Asterisk component that reports security events. However, SIP support will be
1588 coming soon. For more information on the security events framework, see the
1589 "Asterisk Security Framework" section of the Asterisk wiki at
1590 https://wiki.asterisk.org/wiki/x/wgBQ
1591 * SIP support was added in Asterisk 10
1592 * This API now supports IPv6 addresses
1596 * A technology independent fax frontend (res_fax) has been added to Asterisk.
1597 * A spandsp based fax backend (res_fax_spandsp) has been added.
1598 * The app_fax module has been deprecated in favor of the res_fax module and
1599 the new res_fax_spandsp backend.
1600 * The SendFAX and ReceiveFAX applications now send their log messages to a
1601 'fax' logger level, instead of to the generic logger levels. To see these
1602 messages, the system's logger.conf file will need to direct the 'fax' logger
1603 level to one or more destinations; the logger.conf.sample file includes an
1604 example of how to do this. Note that if the 'fax' logger level is *not*
1605 directed to at least one destination, log messages generated by these
1606 applications will be lost, and that if the 'fax' logger level is directed to
1607 the console, the 'core set verbose' and 'core set debug' CLI commands will
1608 have no effect on whether the messages appear on the console or not.
1612 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
1613 Now, in order to enable transmitting silence during record the transmit_silence
1614 option should be used. transmit_silence_during_record remains a valid option, but
1615 defaults to the behavior of the transmit_silence option.
1616 * Addition of the Unit Test Framework API for managing registration and execution
1617 of unit tests with the purpose of verifying the operation of C functions.
1618 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
1619 XMPP text messages to the remote JID.
1620 * Modules.conf has a new option - "require" - that marks a module as critical for
1621 the execution of Asterisk.
1622 If one of the required modules fail to load, Asterisk will exit with a return
1624 * An 'X' option has been added to the asterisk application which enables #exec support.
1625 This allows #exec to be used in asterisk.conf.
1626 * jabber.conf supports a new option auth_policy that toggles auto user registration.
1627 * A new lockconfdir option has been added to asterisk.conf to protect the
1628 configuration directory (/etc/asterisk by default) during reloads.
1629 * The parkeddynamic option has been added to features.conf to enable the creation
1630 of dynamic parkinglots.
1631 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
1632 the reportalarms config option.
1633 * chan_dahdi supports dialing configuring and dialing by device file name.
1634 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
1635 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
1636 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
1637 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
1638 Handy for the above name-based syntax as it does not depend on
1639 initialization order.
1640 * The Realtime dialplan switch now caches entries for 1 second. This provides a
1641 significant increase in performance (about 3X) for installations using this switchtype.
1642 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
1643 AIS. For more information, please see the Distributed Device State section of the
1644 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1645 * The addition of G.719 pass-through support.
1646 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
1647 during device configuration.
1648 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
1649 have less than 3 lines on the LCD.
1650 * Realtime now supports database failover. See the sample extconfig.conf for details.
1651 * The addition of improved translation path building for wideband codecs. Sample
1652 rate changes during translation are now avoided unless absolutely necessary.
1653 * The addition of the res_stun_monitor module for monitoring and reacting to network
1654 changes while behind a NAT.
1655 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
1656 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
1657 These allow support for any Administration. Default is AT&T values.
1661 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
1662 optionally accept a filename, to apply the setting only to the code generated from
1663 that source file when Asterisk was built. However, there are some modules in Asterisk
1664 that are composed of multiple source files, so this did not result in the behavior
1665 that users expected. In this version, 'core set debug' and 'core set verbose'
1666 can optionally accept *module* names instead (with or without the .so extension),
1667 which applies the setting to the entire module specified, regardless of which source
1668 files it was built from.
1669 * New 'manager show settings' command showing the current settings loaded from
1671 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
1672 the channel hangup request to all channels.
1673 * Added a "core reload" CLI command that executes a global reload of Asterisk.
1675 ------------------------------------------------------------------------------
1676 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
1677 ------------------------------------------------------------------------------
1681 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
1682 Snom phones use this for call pickup of extensions that the phone is
1684 * Added support for setting the domain in the URI for caller of an
1685 outbound call by using the SIPFROMDOMAIN channel variable.
1686 * Added a new configuration option "remotesecret" for authentication to
1687 remote services. For backwards compatibility, "secret" still has the
1688 same function as before, but now you can configure both a remote secret and a
1689 local secret for mutual authentication.
1690 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
1691 the sound will be played to the target of an attended transfer
1692 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
1693 finer control over how many peers Asterisk will qualify and the gap between them
1694 when all peers need to be qualified at the same time.
1695 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
1696 (either globally or for a specific peer), chan_sip will treat any SDP data
1697 it receives as new data and update the media stream accordingly. By
1698 default, Asterisk will only modify the media stream if the SDP session
1699 version received is different from the current SDP session version. This
1700 option is required to interoperate with devices that have non-standard SDP
1701 session version implementations (observed with Microsoft OCS). This option
1702 is disabled by default.
1703 * The parsing of register => lines in sip.conf has been modified to allow a port
1704 to be present in the "user" portion. Please see the sip.conf.sample file for more
1706 * Added support for subscribing to MWI on a remote server and making the status available
1707 as a mailbox. Please see the sip.conf.sample file for more information.
1708 * Added a function to remove SIP headers added in the dialplan before the
1709 first INVITE is generated - SIPRemoveHeader()
1710 * Channel variables set with setvar= in a device configuration is now
1711 set both for inbound and outbound calls.
1712 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
1716 * Added immediate option to iax.conf
1717 * Added forceencryption option to iax.conf
1718 * Added Encryption and Trunk status to manager command "iaxpeers"
1722 * The configuration file now holds separate sections for devices and lines.
1723 Please have a look at configs/skinny.conf.sample and change your skinny.conf
1728 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
1729 support for LibOpenR2. http://www.libopenr2.org/
1730 * The UK option waitfordialtone has been added for use with BT analog
1732 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
1733 is used in conjunction with the 'faxdetect' configuration option. When
1734 'faxbuffers' is used and fax tones are detected, the channel will dynamically
1735 switch to the configured faxbuffers policy. For example, to use 6 buffers
1736 and a 'full' buffer policy for a fax transmission, add:
1738 The faxbuffers configuration will be in affect until the call is torn down.
1739 * Added service message support for 4ESS/5ESS switches.
1743 * For DAHDI channels, the CHANNEL() dialplan function now
1744 supports changing the channel's buffer policy (for the current
1745 call only), using this syntax:
1747 exten => s,n,Set(CHANNEL(buffers)=6,full)
1749 This would change the channel to the 'full' buffer policy and
1750 6 (six) buffers. Possible options for this setting are the same
1751 as those in chan_dahdi.conf.
1752 * Added a new dialplan function, CURLOPT, which permits setting various
1753 options that may be useful with the CURL dialplan function, such as
1754 cookies, proxies, connection timeouts, passwords, etc.
1755 * Permit the syntax and synopsis fields of the corresponding dialplan
1756 functions to be individually set from func_odbc.conf.
1757 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
1758 * func_odbc now may specify an insert query to execute, when the write query
1759 affects 0 rows (usually indicating that no such row exists).
1760 * Added a new dialplan function, LISTFILTER, which permits removing elements
1761 from a set list, by name. Uses the same general syntax as the existing CUT
1762 and FIELDQTY dialplan functions, which also manage lists.
1763 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
1764 obtaining realtime data from the dialplan.
1765 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
1766 a subroutine when using the GoSub() and Return() applications.
1767 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
1768 of "core show function AUDIOHOOK_INHERIT" from the CLI
1769 * Added AES_ENCRYPT. For information on its use, please see the output
1770 of "core show function AES_ENCRYPT" from the CLI
1771 * Added AES_DECRYPT. For information on its use, please see the output
1772 of "core show function AES_DECRYPT" from the CLI
1773 * func_odbc now supports database transactions across multiple queries.
1777 * Scheduled meetme conferences may now have their end times extended by
1779 * app_authenticate now gives the ability to select a prompt other than
1781 * app_directory now pays attention to the searchcontexts setting in
1782 voicemail.conf and will look through all contexts, if no context is
1783 specified in the initial argument.
1784 * A new application, Originate, has been introduced, that allows asynchronous
1785 call origination from the dialplan.
1786 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
1787 in addition to the setting in the "general" context.
1788 * Added ConfBridge dialplan application which does conference bridges without
1789 DAHDI. For information on its use, please see the output of
1790 "core show application ConfBridge" from the CLI.
1794 * The Asterisk CLI has a new command, "channel redirect", which is similar in
1795 operation to the AMI Redirect action.
1796 * extensions.conf now allows you to use keyword "same" to define an extension
1797 without actually specifying an extension. It uses exactly the same pattern
1798 as previously used on the last "exten" line. For example:
1799 exten => 123,1,NoOp(something)
1800 same => n,SomethingElse()
1801 * musiconhold.conf classes of type 'files' can now use relative directory paths,
1802 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
1803 * All deprecated CLI commands are removed from the sourcecode. They are now handled
1804 by the new clialiases module. See cli_aliases.conf.sample file.
1805 * Times within timespecs are now accurate down to the minute. This is a change
1806 from historical Asterisk, which only provided timespecs rounded to the nearest
1807 even (read: evenly divisible by 2) minute mark.
1808 * The realtime switch now supports an option flag, 'p', which disables searches for
1810 * In addition to a time range and date range, timespecs now accept a 5th optional
1811 argument, timezone. This allows you to perform time checks on alternate
1812 timezones, especially if those daylight savings time ranges vary from your
1813 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
1815 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
1816 give you the correct output for an asterisk box behind nat. It will give you the
1817 externhost and localnet settings.
1818 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
1819 can connect calls in passthrough mode, as well as record and play back files.
1820 * Successful and unsuccessful call pickup can now be alerted through sounds, by
1821 using pickupsound and pickupfailsound in features.conf.
1822 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
1823 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
1824 instead of the /var/run/asterisk.pid where it used to be. This will make
1825 installs as non-root easier to manage.
1830 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
1831 be written; they will no longer be explicitly written.
1833 Asterisk Manager Interface
1834 --------------------------
1835 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
1836 a non-empty value) in your request. If you do this, any pending AMI events will
1837 *not* be included in the response to your request as they would normally, but
1838 will be left in the event queue for the next request you make to retrieve. For
1839 some applications, this will allow you to guarantee that you will only see
1840 events in responses to 'WaitEvent' actions, and can better know when to expect them.
1841 To know whether the Asterisk server supports this header or not, your client can
1842 inspect the first response back from the server to see if it includes this header:
1844 Pragma: SuppressEvents
1846 If this is included, the server supports event suppression.
1848 * Added 4 new Actions to list skinny device(s) and line(s)
1854 LDAP Schema File Additions
1855 --------------------------
1856 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
1857 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
1859 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
1860 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
1861 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
1862 * Removed redundant IPaddr (there's already IPAddress)
1863 - Gives more configuration Flags for SIP-Users available (tested)
1864 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
1865 without extensibleObject (which really should be the last resort); gives
1866 also additional possibilities for LDAP-filter
1868 ------------------------------------------------------------------------------
1869 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
1870 ------------------------------------------------------------------------------
1872 Device State Handling
1873 ---------------------
1874 * The event infrastructure in Asterisk got another big update to help support
1875 distributed events. It currently supports distributed device state and
1876 distributed Voicemail MWI (Message Waiting Indication). A new module has
1877 been merged, res_ais, which facilitates communicating events between servers.
1878 It uses the SAForum AIS (Service Availability Forum Application Interface
1879 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
1880 a cluster of Asterisk servers, and to share events between them. For more
1881 information on setting this up, refer to the Distributed Device State section
1882 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1886 * Added a new dialplan function, AST_CONFIG(), which allows you to access
1887 variables from an Asterisk configuration file.
1888 * The JACK_HOOK function now has a c() option to supply a custom client name.
1889 * Added two new dialplan functions from libspeex for audio gain control and
1890 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
1891 rx directions of a channel from the dialplan.
1892 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
1893 based on other parameters. The default is still to search based on the
1894 forwarding station ID. However, there are new options that allow you to search
1895 based on the message desk terminal ID, or the message desk number.
1896 * TIMEOUT() has been modified to be accurate down to the millisecond.
1897 * ENUM*() functions now include the following new options:
1898 - 'u' returns the full URI and does not strip off the URI-scheme.
1899 - 's' triggers ISN specific rewriting
1900 - 'i' looks for branches into an Infrastructure ENUM tree
1901 - 'd' for a direct DNS lookup without any flipping of digits.
1902 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
1903 * CHANNEL() now has options for the maximum, minimum, and standard or normal
1904 deviation of jitter, rtt, and loss for a call using chan_sip.
1906 DAHDI channel driver (chan_dahdi) Changes
1907 ----------------------------------------
1908 * Channels can now be configured using named sections in chan_dahdi.conf, just
1909 like other channel drivers, including the use of templates.
1910 * The default for pridialplan has changed from 'national' to 'unknown'.
1914 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
1915 to something that matches the pattern a hint will be created using the contents
1916 and variables evaluated.
1917 * Dialplan matching has been extended to allow an extension to return to the
1918 PBX core to wait for more digits. This is done by using the new dialplan
1919 application called "Incomplete". This will permit a whole new level of
1920 extension control, by giving the administrator more control over early
1921 matches employing one of the short-circuit pattern match operators. Note
1922 that custom applications can trigger this same behavior by returning the
1923 special value AST_PBX_INCOMPLETE.
1927 * Directory now permits both first and last names to be matched at the same
1928 time. In addition, the number of digits to enter of the name can be set in
1929 the arguments to Directory; previously, you could enter only 3, regardless
1930 of how many names are in your company. For large companies, this should be
1932 * Voicemail now permits a mailbox setting to wrap around from first to last
1933 messages, if the "messagewrap" option is set to a true value.
1934 * Voicemail now permits an external script to be run, for password validation.
1935 The script should output "VALID" or "INVALID" on stdout, depending upon the
1936 wish to validate or invalidate the password given. Arguments are:
1937 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
1939 * Dial has a new option: F(context^extension^pri), which permits a callee to
1940 continue in the dialplan, at the specified label, if the caller hangs up.
1941 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
1942 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
1943 * The Jack application now has a c() option to supply a custom client name.
1944 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
1945 like the pre-existing whisper mode, except that the spy can also talk to the
1946 participant on the bridged channel as well.
1947 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
1948 to be spoken instead of the channel name or number. For more information on the
1949 use of this option, issue the command "core show application ChanSpy" from the
1951 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
1952 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
1953 words, if using the 'd' option, it is not possible to enter a number to append to
1954 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
1955 change to whisper mode, and pressing 6 will change to barge mode.
1956 * ExternalIVR now takes several options that affect the way it performs, as
1957 well as having several new commands. Please see the External IVR page on the Asterisk
1958 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
1959 * Added ability to communicate over a TCP socket instead of forking a child process for the
1960 ExternalIVR application.
1961 * ChanIsAvail has a new option, 'a', which will return all available channels instead
1962 of just the first one if you give the function more then one channel to check.
1963 * PrivacyManager now takes an option where you can specify a context where the
1964 given number will be matched. This way you have more control over who is allowed
1965 and it stops the people who blindly enter 10 digits.
1966 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
1967 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
1968 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
1969 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
1970 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
1971 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
1972 * The Dial() application no longer copies the language used by the caller to the callee's
1973 channel. If you desire for the caller's channel's language to be used for file playback
1974 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
1975 * SendImage() no longer hangs up the channel on error; instead, it sets the
1976 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
1977 'UNSUPPORTED'. This change makes SendImage() more consistent with other
1979 * Park has a new option, 's', which silences the announcement of the parking space number.
1980 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
1981 invalid input and will be assumed to mean that no timeout is desired.
1985 * Added DNS manager support to registrations for peers referencing peer entries.
1986 DNS manager runs in the background which allows DNS lookups to be run asynchronously
1987 as well as periodically updating the IP address. These properties allow for
1988 better performance as well as recovery in the event of an IP change.
1989 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
1990 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
1991 These changes also provide performance improvements for call setup and tear down.
1992 * Added ability to specify registration expiry time on a per registration basis in
1994 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
1996 * Added t38pt_usertpsource option. See sip.conf.sample for details.
1997 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
1998 * 'sip show peers' and 'sip show users' display their entries sorted in
1999 alphabetical order, as opposed to the order they were in, in the config
2001 * Videosupport now supports an additional option, "always", which always sets
2002 up video RTP ports, even on clients that don't support it. This helps with
2003 callfiles and certain transfers to ensure that if two video phones are
2004 connected, they will always share video feeds.
2008 * Existing DNS manager lookups extended to check for SRV records.
2009 * IAX2 encryption support has been improved to support periodic key rotation
2010 within a call for enhanced security. The option "keyrotate" has been
2011 provided to disable this functionality to preserve backwards compatibility
2012 with older versions of IAX2 that do not support key rotation.
2016 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
2017 data tree based on the given <path>.
2018 * New CLI command "data show providers" that will display all the registered
2020 * New CLI command, "config reload <file.conf>" which reloads any module that
2021 references that particular configuration file. Also added "config list"
2022 which shows which configuration files are in use.
2023 * New CLI commands, "pri show version" and "ss7 show version" that will
2024 display which version of libpri and libss7 are being used, respectively.
2025 A new API call was added so trunk will now have to be compiled against
2026 a versions of libpri and libss7 that have them or it will not know that
2027 these libraries exist.
2028 * The commands "core show globals", "core set global" and "core set chanvar" has
2029 been deprecated in favor of the more semanticly correct "dialplan show globals",
2030 "dialplan set chanvar" and "dialplan set global".
2031 * New CLI command "dialplan show chanvar" to list all variables associated
2032 with a given channel.
2036 * Addresses managed by DNS manager now can check to see if there is a DNS
2037 SRV record for a given domain and will use that hostname/port if present.
2039 AMI - The manager (TCP/TLS/HTTP)
2040 --------------------------------
2041 * The Status command now takes an optional list of variables to display
2042 along with channel status.
2043 * The QueueEntry event now also includes the channel's uniqueid
2047 * res_odbc no longer has a limit of 1023 total possible unshared connections,
2048 as some people were running into this limit. This limit has been increased
2053 * The TRANSFER queue log entry now includes the the caller's original
2054 position in the transferred-from queue.
2055 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
2056 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
2057 as well as an explanation about timeout options in general
2058 * Added a new option - C - for forcing the "answered elsewhere" flag on
2059 cancellation of calls in to members of the queue. This is to avoid the
2060 call to a member of a queue having the call listed as a "missed call".
2064 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
2065 adaptive capabilities. What this means in practical terms is that if your
2066 realtime table lacks critical fields, Asterisk will now emit warnings to
2067 that effect. Also, some of the realtime drivers have the ability (if
2068 configured) to automatically add those columns to the table with the
2069 correct type and length.
2073 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
2074 the 'setvar' option to cause a given audio file to be played upon completion
2075 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
2076 Skinny channels only.
2077 * You can now compile Asterisk against the Hoard Memory Allocator, see the
2078 Hoard page on the Asterisk wiki for more information:
2079 https://wiki.asterisk.org/wiki/x/pQBB
2080 * Config file variables may now be appended to, by using the '+=' append
2081 operator. This is most helpful when working with long SQL queries in
2082 func_odbc.conf, as the queries no longer need to be specified on a single
2084 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
2085 which will add a second to the billsec when the ending
2086 time is set, if the number in the microseconds field of the end time is
2087 greater than the number of microseconds in the answer time. This allows
2088 users to count the 'initiated' seconds in their billing records.
2090 ------------------------------------------------------------------------------
2091 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
2092 ------------------------------------------------------------------------------
2094 AMI - The manager (TCP/TLS/HTTP)
2095 --------------------------------
2096 * Manager has undergone a lot of changes, all of them documented
2097 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
2098 * Manager version has changed to 1.1
2099 * Added a new action 'CoreShowChannels' to list currently defined channels
2100 and some information about them.
2101 * Added a new action 'SIPshowregistry' to list SIP registrations.
2102 * Added TLS support for the manager interface and HTTP server
2103 * Added the URI redirect option for the built-in HTTP server
2104 * The output of CallerID in Manager events is now more consistent.
2105 CallerIDNum is used for number and CallerIDName for name.
2106 * Enable https support for builtin web server.
2107 See configs/http.conf.sample for details.
2108 * Added a new action, GetConfigJSON, which can return the contents of an
2109 Asterisk configuration file in JSON format. This is intended to help
2110 improve the performance of AJAX applications using the manager interface
2112 * SIP and IAX manager events now use "ChannelType" in all cases where we
2113 indicate channel driver. Previously, we used a mixture of "Channel"
2114 and "ChannelDriver" headers.
2115 * Added a "Bridge" action which allows you to bridge any two channels that
2116 are currently active on the system.
2117 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
2118 the voicemail users setup.
2119 * Added 'DBDel' and 'DBDelTree' manager commands.
2120 * cdr_manager now reports events via the "cdr" level, separating it from
2121 the very verbose "call" level.
2122 * Manager users are now stored in memory. If you change the manager account
2123 list (delete or add accounts) you need to reload manager.
2124 * Added Masquerade manager event for when a masquerade happens between
2126 * Added "manager reload" command for the CLI
2127 * Lots of commands that only provided information are now allowed under the
2128 Reporting privilege, instead of only under Call or System.
2129 * The IAX* commands now require either System or Reporting privilege, to
2130 mirror the privileges of the SIP* commands.
2131 * Added ability to retrieve list of categories in a config file.
2132 * Added ability to retrieve the content of a particular category.
2133 * Added ability to empty a context.
2134 * Created new action to create a new file.
2135 * Updated delete action to allow deletion by line number with respect to category.
2136 * Added new action insert to add new variable to category at specified line.
2137 * Updated action newcat to allow new category to be inserted in file above another
2139 * Added new event "JitterBufStats" in the IAX2 channel
2140 * Originate now requires the Originate privilege and, if you want to call out
2141 to a subshell, it requires the System privilege, as well. This was done to
2142 enhance manager security.
2143 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
2144 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
2145 or manager show command Atxfer from the CLI
2146 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
2147 details or manager show command IAXregistry from the CLI
2151 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
2152 state in the dialplan, as well as creating custom device states that are
2153 controllable from the dialplan.
2154 * Extend CALLERID() function with "pres" and "ton" parameters to
2155 fetch string representation of calling number presentation indicator
2156 and numeric representation of type of calling number value.
2157 * MailboxExists converted to dialplan function
2158 * A new option to Dial() for telling IP phones not to count the call
2159 as "missed" when dial times out and cancels.
2160 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
2161 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
2162 held for any given channel. Also, locks are automatically freed when a
2164 * Added HINT() dialplan function that allows retrieving hint information.
2165 Hints are mappings between extensions and devices for the sake of
2166 determining the state of an extension. This function can retrieve the list
2167 of devices or the name associated with a hint.
2168 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
2170 * Added SYSINFO() dialplan function which allows retrieval of system information
2171 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
2172 the existence of a dialplan target.
2173 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
2174 upper and lower case, respectively.
2175 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
2176 ID for the call (not the Asterisk call ID or unique ID), provided that the
2177 channel driver supports this. For SIP, you get the SIP call-ID for the
2178 bridged channel which you can store in the CDR with a custom field.
2182 * Added CLI permissions, config file: cli_permissions.conf
2183 default is to allow all commands for every local user/group.
2184 Also this new feature added three new CLI commands:
2185 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
2186 - cli reload permissions
2187 - cli show permissions
2188 * New CLI command "core show hint" (usage: core show hint <exten>)
2189 * New CLI command "core show settings"
2190 * Added 'core show channels count' CLI command.
2191 * Added the ability to set the core debug and verbose values on a per-file basis.
2192 * Added 'queue pause member' and 'queue unpause member' CLI commands
2193 * Ability to set process limits ("ulimit") without restarting Asterisk
2194 * Enhanced "agi debug" to print the channel name as a prefix to the debug
2195 output to make debugging on busy systems much easier.
2196 * New CLI commands "dialplan set extenpatternmatching true/false"
2197 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
2198 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
2199 listed in the startup_commands section of cli.conf will get executed.
2200 * Added a CLI command, "devstate change", which allows you to set custom device
2201 states from the func_devstate module that provides the DEVICE_STATE() function
2202 and handling of the "Custom:" devices.
2203 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
2204 sorted into the different possible callbacks, with the number of entries
2205 currently scheduled for each. Gives you a feel for how busy the sip channel
2207 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
2208 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
2209 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
2213 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
2214 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
2215 for a received call. If it is detected, the channel will jump to the
2216 'fax' extension in the dialplan.
2217 * The default SIP useragent= identifier now includes the Asterisk version
2218 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
2219 If set, and the incoming request carries authentication info,
2220 the username to match in the users list is taken from the Digest header
2221 rather than from the From: field. This feature is considered experimental.
2222 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
2223 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
2224 * The "localmask" setting was removed in version 1.2 and the reminder about it
2225 being removed is now also removed.
2226 * A new option "busylevel" for setting a level of calls where asterisk reports
2227 a device as busy, to separate it from call-limit. This value is also added
2228 to the SIP_PEER dialplan function.
2229 * A new realtime family called "sipregs" is now supported to store SIP registration
2230 data. If this family is defined, "sippeers" will be used for configuration and
2231 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
2232 registration data, as before.
2233 * The SIPPEER function have new options for port address, call and pickup groups
2234 * Added support for T.140 realtime text in SIP/RTP
2235 * The "checkmwi" option has been removed from sip.conf, as it is no longer
2236 required due to the restructuring of how MWI is handled. See the descriptions
2237 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
2238 for more information.
2239 * Added rtpdest option to CHANNEL() dialplan function.
2240 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
2241 * SIP now adds a header to the CANCEL if the call was answered by another phone
2242 in the same dial command, or if the new c option in dial() is used.
2243 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
2244 states it is not needed. For phones, however, that do require it the "registertrying" option
2245 has been added so it can be enabled.
2246 * A new option called "callcounter" (global/peer/user level) enables call counters needed
2247 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
2248 used to enable this functionality).
2249 * New settings for timer T1 and timer B on a global level or per device. This makes it
2250 possible to force timeout faster on non-responsive SIP servers. These settings are
2251 considered advanced, so don't use them unless you have a problem.
2252 * Added a dial string option to be able to set the To: header in an INVITE to any
2254 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
2255 the qualify frequency.
2256 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
2257 were not properly torn down due to network or endpoint failures during an established
2259 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
2260 and configs/sip.conf.sample for more information on how it is used.
2261 * Added a new configuration option "authfailureevents" that enables manager events when
2262 a peer can't authenticate properly.
2263 * Added DNS manager support to registrations for peers not referencing a peer entry.
2267 * Added the trunkmaxsize configuration option to chan_iax2.
2268 * Added the srvlookup option to iax.conf
2269 * Added support for OSP. The token is set and retrieved through the CHANNEL()
2272 XMPP Google Talk/Jingle changes
2273 -------------------------------
2274 * Added the bindaddr option to gtalk.conf.
2278 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
2279 * Proper codec support in chan_skinny.
2280 * Added settings for IP and Ethernet QoS requests
2284 * Added separate settings for media QoS in mgcp.conf
2286 Console Channel Driver changes
2287 ------------------------------
2288 * Added experimental support for video send & receive to chan_oss.
2289 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
2292 Phone channel changes (chan_phone)
2293 ----------------------------------
2294 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
2296 H.323 channel Changes
2297 ---------------------
2298 * H323 remote hold notification support added (by NOTIFY message
2299 and/or H.450 supplementary service)
2301 Local channel changes
2302 ---------------------
2303 * The device state functionality in the Local channel driver has been updated
2304 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
2305 to just UNKNOWN if the extension exists.
2306 * Added jitterbuffer support for chan_local. This allows you to use the
2307 generic jitterbuffer on incoming calls going to Asterisk applications.
2308 For example, this would allow you to use a jitterbuffer for an incoming
2309 SIP call to Voicemail by putting a Local channel in the middle. This
2310 feature is enabled by using the 'j' option in the Dial string to the Local
2311 channel in conjunction with the existing 'n' option for local channels.
2312 * A 'b' option has been added which causes chan_local to return the actual channel
2313 that is behind it when queried. This is useful for transfer scenarios as the
2314 actual channel will be transferred, not the Local channel.
2316 Agent channel changes
2317 ----------------------
2318 * The ackcall and endcall options are now supplemented with options acceptdtmf
2319 and enddtmf. These allow for the DTMF keypress to be configurable. The options
2320 default to their old hard-coded values ('#' and '*' respectively) so this should
2321 not break any existing agent installations.
2323 DAHDI channel driver (chan_dahdi) Changes
2324 ----------------------------------------
2325 * SS7 support (via libss7 library)
2326 * In India, some carriers transmit CID via dtmf. Some code has been added
2327 that will handle some situations. The cidstart=polarity_IN choice has been added for
2328 those carriers that transmit CID via dtmf after a polarity change.
2329 * CID matching information is now shown when doing 'dialplan show'.
2330 * Added dahdi show version CLI command.
2331 * Added setvar support to chan_dahdi.conf channel entries.
2332 * Added two new options: mwimonitor and mwimonitornotify. These options allow
2333 you to enable MWI monitoring on FXO lines. When the MWI state changes,
2334 the script specified in the mwimonitornotify option is executed. An internal
2335 event indicating the new state of the mailbox is also generated, so that
2336 the normal MWI facilities in Asterisk work as usual.
2337 * Added signalling type 'auto', which attempts to use the same signalling type
2338 for a channel as configured in DAHDI. This is primarily designed for analog
2339 ports, but will also work for digital ports that are configured for FXS or FXO
2340 signalling types. This mode is also the default now, so if your chan_dahdi.conf
2341 does not specify signalling for a channel (which is unlikely as the sample
2342 configuration file has always recommended specifying it for every channel) then
2343 the 'auto' mode will be used for that channel if possible.
2344 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
2345 state for a channel; also ensured that the DNDState Manager event is
2346 emitted no matter how the DND state is set or cleared.
2350 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
2351 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
2352 for details. This new channel driver allows you to use Nortel i2002,
2353 i2004, and i2050 phones with Asterisk.
2354 * Added a new channel driver, chan_console, which uses portaudio as a cross
2355 platform audio interface. It was written as a channel driver that would
2356 work with Mac CoreAudio, but portaudio supports a number of other audio
2357 interfaces, as well. Note that this channel driver requires v19 or higher
2358 of portaudio; older versions have a different API.
2362 * Added the ability to specify arguments to the Dial application when using
2363 the DUNDi switch in the dialplan.
2364 * Added the ability to set weights for responses dynamically. This can be
2365 done using a global variable or a dialplan function. Using the SHELL()
2366 function would allow you to have an external script set the weight for
2368 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
2369 functions will allow you to initiate a DUNDi query from the dialplan,
2370 find out how many results there are, and access each one.
2371 * Added the ability to specifiy a port for a dundi peer.
2375 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
2376 functions will allow you to initiate an ENUM lookup from the dialplan,
2377 and Asterisk will cache the results. ENUMRESULT can be used to access
2378 the results without doing multiple DNS queries.
2382 * Added the ability to customize which sound files are used for some of the
2383 prompts within the Voicemail application by changing them in voicemail.conf
2384 * Added the ability for the "voicemail show users" CLI command to show users
2385 configured by the dynamic realtime configuration method.
2386 * MWI (Message Waiting Indication) handling has been significantly
2387 restructured internally to Asterisk. It is now totally event based
2388 instead of polling based. The voicemail application will notify other
2389 modules that have subscribed to MWI events when something in the mailbox
2391 This also means that if any other entity outside of Asterisk is changing
2392 the contents of mailboxes, then the voicemail application still needs to
2393 poll for changes. Examples of situations that would require this option
2394 are web interfaces to voicemail or an email client in the case of using
2395 IMAP storage. So, two new options have been added to voicemail.conf
2396 to account for this: "pollmailboxes" and "pollfreq". See the sample
2397 configuration file for details.
2398 * Added "tw" language support
2399 * Added support for storage of greetings using an IMAP server
2400 * Added ability to customize forward, reverse, stop, and pause keys for message playback
2401 * SMDI is now enabled in voicemail using the smdienable option.
2402 * A "lockmode" option has been added to asterisk.conf to configure the file
2403 locking method used for voicemail, and potentially other things in the
2404 future. The default is the old behavior, lockfile. However, there is a
2405 new method, "flock", that uses a different method for situations where the
2406 lockfile will not work, such as on SMB/CIFS mounts.
2407 * Added the ability to backup deleted messages, to ease recovery in the case
2408 that a user accidentally deletes a message, and discovers that they need it.
2409 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
2410 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
2411 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
2412 voicemail boxes. The SMDI interface can also poll for MWI changes when some
2413 outside entity is modifying the state of the mailbox (such as IMAP storage or
2414 a web interface of some kind).
2415 * Added the support for marking messages as "urgent." There are two methods to accomplish
2416 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
2417 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
2418 the message as urgent after he has recorded a voicemail by following the voice instructions.
2419 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
2424 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
2425 used across multiple queues.
2426 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
2427 setqueueentryvar options for each queue, see queues.conf.sample for details.
2428 * Added keepstats option to queues.conf which will keep queue
2429 statistics during a reload.
2430 * setinterfacevar option in queues.conf also now sets a variable
2431 called MEMBERNAME which contains the member's name.
2432 * Added 'Strategy' field to manager event QueueParams which represents
2433 the queue strategy in use.
2434 * Added option to run macro when a queue member is connected to a caller,
2435 see queues.conf.sample for details.
2436 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
2437 does not count paused queue members as unavailable.
2438 * Added min-announce-frequency option to queues.conf which allows you to control the
2439 minimum amount of time between queue announcements for use when the caller's queue
2440 position changes frequently.
2441 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
2443 * Added ability for non-realtime queues to have realtime members
2444 * Added the "linear" strategy to queues.
2445 * Added the "wrandom" strategy to queues.
2446 * Added new channel variable QUEUE_MIN_PENALTY
2447 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
2448 rules in queuerules.conf. See configs/queuerules.conf.sample for details
2449 * Added a new parameter for member definition, called state_interface. This may be
2450 used so that a member may be called via one interface but have a different interface's
2451 device state reported.
2452 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
2453 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
2454 "manager show command QueueReset."
2455 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
2456 specified by the periodic-announce option, then one will be chosen randomly when it is time
2457 to play a periodic announcment
2458 * New configuration options: announce-position now takes two more values in addition to "yes" and
2459 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
2460 announce-position-limit. By setting announce-position to "limit" callers will only have their
2461 position announced if their position is less than what is specified by announce-position-limit.
2462 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
2463 will be told that their are more than announce-position-limit callers waiting.
2464 * Two new queue log events have been added. An ADDMEMBER event will be logged
2465 when a realtime queue member is added and a REMOVEMEMBER event will be logged
2466 when a realtime queue member is removed. Since there is no calling channel associated
2467 with these events, the string "REALTIME" is placed where the channel's unique id
2468 is typically placed.
2469 * The configuration method for the "joinempty" and "leavewhenempty" options has
2470 changed to a comma-separated list of methods of determining member availability
2471 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
2472 values are still accepted for backwards-compatibility, though.
2473 * The average talktime is now calculated on queues. This information is reported via the
2474 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
2475 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
2480 * The 'o' option to provide an optimization has been removed and its functionality
2481 has been enabled by default.
2482 * When a conference is created, the UNIQUEID of the channel that caused it to be
2483 created is stored. Then, every channel that joins the conference will have the
2484 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
2485 callers that come and go from long standing conferences.
2486 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
2487 except it does operations on a channel by name, instead of number in a conference.
2488 This is a very useful feature in combination with the 'X' option to ChanSpy.
2489 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
2491 * Added new RealTime functionality to provide support for scheduled conferencing.
2492 This includes optional messages to the caller if they attempt to join before
2493 the schedule start time, or to allow the caller to join the conference early.
2494 Also included is optional support for limiting the number of callers per
2495 RealTime conference.
2496 * Added the S() and L() options to the MeetMe application. These are pretty
2497 much identical to the S() and L() options to Dial(). They let you set
2498 timeouts for the conference, as well as have warning sounds played to
2499 let the caller know how much time is left, and when it is running out.
2500 * Added the ability to do "meetme concise" with the "meetme" CLI command.
2501 This extends the concise capabilities of this CLI command to include
2502 listing all conferences, instead of an addition to the other sub commands
2503 for the "meetme" command.
2504 * Added the ability to specify the music on hold class used to play into the
2505 conference when there is only one member and the M option is used.
2506 * Added MEETME_INFO dialplan function which provides a way to query
2507 various properties of a Meetme conference.
2508 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
2509 and *84: record in-conf
2511 Other Dialplan Application Changes
2512 ----------------------------------
2513 * Argument support for Gosub application
2514 * From the to-do lists: straighten out the app timeout args:
2515 Wait() app now really does 0.3 seconds- was truncating arg to an int.
2516 WaitExten() same as Wait().
2517 Congestion() - Now takes floating pt. argument.
2518 Busy() - now takes floating pt. argument.
2519 Read() - timeout now can be floating pt.
2520 WaitForRing() now takes floating pt timeout arg.
2521 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
2522 * Added 's' option to Page application.
2523 * Added an optional timeout argument to the Page application.
2524 * Added 'E', 'V', and 'P' commands to ExternalIVR.
2525 * Added 'o' and 'X' options to Chanspy.
2526 * Added a new dialplan application, Bridge, which allows you to bridge the
2527 calling channel to any other active channel on the system.
2528 * Added the ability to specify a music on hold class to play instead of ringing
2529 for the SLATrunk application.
2530 * The Read application no longer exits the dialplan on error. Instead, it sets
2531 READSTATUS to ERROR, which you can catch and handle separately.
2532 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
2533 of asking for verification of each name, one at a time.
2534 * Privacy() no longer uses privacy.conf, as all options are specifyable as
2535 direct options to the app.
2536 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
2538 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
2539 * The ChannelRedirect application no longer exits the dialplan if the given channel
2540 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
2541 or NOCHANNEL if the given channel was not found.
2542 * The silencethreshold setting that was previously configurable in multiple
2543 applications is now settable globally via dsp.conf.
2545 Music On Hold Changes
2546 ---------------------
2547 * A new option, "digit", has been added for music on hold classes in
2548 musiconhold.conf. If this is set for a music on hold class, a caller
2549 listening to music on hold can press this digit to switch to listening
2550 to this music on hold class.
2551 * Support for realtime music on hold has been added.
2552 * In conjunction with the realtime music on hold, a general section has
2553 been added to musiconhold.conf, its sole variable is cachertclasses. If this
2554 is set, then music on hold classes found in realtime will be cached in memory.
2558 * AEL upgraded to use the Gosub with Arguments instead
2559 of Macro application, to hopefully reduce the problems
2560 seen with the artificially low stack ceiling that
2561 Macro bumps into. Macros can only call other Macros
2562 to a depth of 7. Tests run using gosub, show depths
2563 limited only by virtual memory. A small test demonstrated
2564 recursive call depths of 100,000 without problems.
2565 -- in addition to this, all apps that allowed a macro
2566 to be called, as in Dial, queues, etc, are now allowing
2567 a gosub call in similar fashion.
2568 * AEL now generates LOCAL(argname) declarations when it
2569 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
2570 etc. That makes the arguments local in scope. The user
2571 can define their own local variables in macros, now,
2572 by saying "local myvar=someval;" or using Set() in this
2573 fashion: Set(LOCAL(myvar)=someval); ("local" is now
2575 * utils/conf2ael introduced. Will convert an extensions.conf
2576 file into extensions.ael. Very crude and unfinished, but
2577 will be improved as time goes by. Should be useful for a
2578 first pass at conversion.
2579 * aelparse will now read extensions.conf to see if a referenced
2580 macro or context is there before issueing a warning.
2581 * AEL parser sets a local channel variable ~~EXTEN~~, to
2582 preserve the value of ${EXTEN} thru switch statements.
2583 * New operator in $[...] expressions: the ~~ operator serves
2584 as a concatenation operator. AT THE MOMENT, it is really only
2585 necessary and useful in AEL, especially in if() expressions.
2586 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
2587 any enclosing double-quotes, and evaluate to the value of a
2588 concatenated with the value of b. For example if a is set to
2589 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
2590 evaluate to xyzabc .
2593 Call Features (res_features) Changes
2594 ------------------------------------
2595 * Added the parkedcalltransfers option to features.conf
2596 * Added parkedcallparking option to control one touch parking w/ parking
2598 * Added parkedcallhangup option to control disconnect feature w/ parking
2600 * Added parkedcallrecording option to control one-touch record w/ parking
2602 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
2603 parkedcalltransfers option support for multiple parking lots.
2604 * Added BRIDGE_FEATURES variable to set available features for a channel
2605 * The built-in method for doing attended transfers has been updated to
2606 include some new options that allow you to have the transferee sent
2607 back to the person that did the transfer if the transfer is not successful.
2608 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
2609 in features.conf.sample.
2610 * Added support for configuring named groups of custom call features in
2611 features.conf. This means that features can be written a single time, and
2612 then mapped into groups of features for different key mappings or easier
2614 * Updated the ParkedCall application to allow you to not specify a parking
2615 extension. If you don't specify a parking space to pick up, it will grab
2616 the first one available.
2617 * Added cli command 'features reload' to reload call features from features.conf
2618 * Moved into core asterisk binary.
2619 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
2620 * Added the ability for custom parking lots to be configured with their own
2621 parking extension with the parkext option.
2623 Language Support Changes
2624 ------------------------
2625 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
2626 * Added support for the Hungarian language for saying numbers, dates, and times.
2630 * Added SPEECH commands for speech recognition. A complete listing can be found
2632 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
2633 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
2634 does not behave as expected; the native command needs to be used, instead.
2635 * Added the ability to perform SRV lookups on fast AGI calls. To use this
2636 feature, simply use hagi: instead of agi: as the protocol portion
2637 of the URI parameter to the AGI function call in your dial plan. Also note
2638 that specifying a port number in the AGI URI will disable SRV lookups,
2639 even if you use the hagi: protocol.
2640 * No longer support MSG_OOB flag on HANGUP.
2644 * Added rotatestrategy option to logger.conf, along with two new options:
2645 "timestamp" which will use the time to name the logger files instead of
2646 sequence number; and "rotate", which rotates the names of the log files,
2647 similar to the way syslog rotates files.
2648 * Added exec_after_rotate option to logger.conf, which allows a system
2649 command to be run after rotation. This is primarily useful with
2650 rotatestrategy=rotate, to allow a limit on the number of log files kept
2651 and to ensure that the oldest log file gets deleted.
2652 * Added realtime support for the queue log
2656 * The cdr_manager module has a [mappings] feature, like cdr_custom,
2657 to add fields to the manager event from the CDR variables.
2658 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
2659 backend database CDR table. Specifically, additional, non-standard
2660 columns are supported, merely by setting the corresponding CDR variable in
2661 your dialplan. In addition, you may alias any column to another name (for
2662 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
2663 simply "alias src => ANI" in the configuration file). Records may be
2664 posted to more than one backend, simply by specifying multiple categories
2665 in the configuration file. And finally, you may filter which CDRs get
2666 posted to each backend, by specifying a filter (which the record must
2667 match) for the particular category. Filters are additive (meaning all
2668 rules must match to post that CDR).
2669 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
2670 module. Specifically, you may add additional columns into the table and
2671 they will be set, if you set the corresponding CDR variable name. Also,
2672 if you omit columns in your database table, they will be silently skipped
2673 (but a record will still be inserted, based on what columns remain). Note
2674 that the other two features from cdr_adaptive_odbc (alias and filter) are
2675 not currently supported.
2676 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
2677 has been disabled using the NoCDR application.
2679 Miscellaneous New Modules
2680 -------------------------
2681 * Added a new CDR module, cdr_sqlite3_custom.
2682 * Added a new realtime configuration module, res_config_sqlite
2683 * Added a new codec translation module, codec_resample, which re-samples
2684 signed linear audio between 8 kHz and 16 kHz to help support wideband
2686 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
2687 based on configuration templates that use Asterisk dialplan function and
2688 variable substitution. It should be possible to create phone profiles and
2689 templates that work for the majority of phones provisioned over http. It
2690 is currently only intended to provision a single user account per phone.
2691 An example profile and set of templates for Polycom phones is provided.
2692 NOTE: Polycom firmware is not included, but should be placed in
2693 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
2694 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
2695 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
2696 provided; there is a JACK() application, and a JACK_HOOK() function. Both
2697 interfaces create an input and output JACK port. The application makes
2698 these ports the endpoint of the call. The audio coming from the channel
2699 goes out the output port and whatever comes back in on the input port is
2700 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
2701 audiohook on the channel. This lets you run the audio coming from a
2702 channel through JACK, and whatever comes back in is what gets forwarded
2703 on as the channel's audio. This is very useful for building custom
2704 vocoders or doing recording or analysis of the channel's audio in another
2706 * Added a new module, res_config_curl, which permits using a HTTP POST url
2707 to retrieve, create, update, and delete realtime information from a remote
2708 web server. Note that this module requires func_curl.so to be loaded for
2709 backend functionality.
2710 * Added a new module, res_config_ldap, which permits the use of an LDAP
2711 server for realtime data access.
2712 * Added support for writing and running your dialplan in lua using the pbx_lua
2713 module. See configs/extensions.lua.sample for examples of how to do this.
2717 * Ability to use libcap to set high ToS bits when non-root
2718 on Linux. If configure is unable to find libcap then you
2719 can use --with-cap to specify the path.
2720 * Added maxfiles option to options section of asterisk.conf which allows you to specify
2721 what Asterisk should set as the maximum number of open files when it loads.
2722 * Added the jittertargetextra configuration option.
2723 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
2724 configuration files for the IP channel drivers. The new option is "cos".
2725 This information is also documented on the Asterisk wiki at
2726 https://wiki.asterisk.org/wiki/x/EYBG
2727 * When originating a call using AMI or pbx_spool that fails the reason for failure
2728 will now be available in the failed extension using the REASON dialplan variable.
2729 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
2730 It allows you to configure a prefix for auto-monitor recordings.
2731 * A new extension pattern matching algorithm, based on a trie, is introduced
2732 here, that could noticeably speed up mid-sized to large dialplans.
2733 It is NOT used by default, as duplicating the behaviour of the old pattern
2734 matcher is still under development. A config file option, in extensions.conf,
2735 in the [general] section, called "extenpatternmatchingnew", is by default
2736 set to false; setting that to true will force the use of the new algorithm.
2737 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
2738 be used to switch the algorithms at run time.
2739 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
2740 specifying which socket to use to connect to the running Asterisk daemon
2742 * Performance enhancements to the sched facility, which is used in
2743 the channel drivers, etc. Added hashtabs and doubly-linked lists
2744 to speed up deletion; start at the beginning or end of list to
2746 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
2747 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
2748 Added regression tests to the tests/ dir, also.
2749 * Added a refcount trace feature to astobj2 for those trying to balance
2750 object creation, deletion; work, play; space and time. See the
2751 notes in astobj2.h. Also, see utils/refcounter as well, as a
2752 quick way to find unbalanced refcounts in what could be a sea
2753 of objects that were balanced.
2754 * Added logging to 'make update' command. See update.log
2755 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
2756 do not come from the remote party.
2757 * Added the 'n' option to the SpeechBackground application to tell it to not
2758 answer the channel if it has not already been answered.
2759 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
2760 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
2762 * iLBC source code no longer included (see UPGRADE.txt for details)
2763 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
2764 deadlock is detected, a backtrace of the stack which led to the lock calls
2765 will be output to the CLI.
2766 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
2767 the "core show locks" CLI command will give lock information output as well
2768 as a backtrace of the stack which led to the lock calls.
2769 * users.conf now sports an optional alternateexts property, which permits
2770 allocation of additional extensions which will reach the specified user.
2771 * A new option for the configure script, --enable-internal-poll, has been added
2772 for use with systems which may have a buggy implementation of the poll system
2773 call. If you notice odd behavior such as the CLI being unresponsive on remote
2774 consoles, you may want to try using this option. This option is enabled by default
2775 on Darwin systems since it is known that the Darwin poll() implementation has
2779 --------------------
2780 * In addition to timing from DAHDI, there is a new timing module called
2781 res_timing_timerfd. In order to use this, you must be running Linux with
2782 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
2783 script will be able to tell if you have the requirements. From menuselect, select
2784 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob