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.
49 * Added general support for busy detection.
51 * Added ECAM command support for Sony Ericsson phones.
55 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
56 using the 'supportpath' setting, either on a global basis or on a peer basis.
57 This setting enables Asterisk to route outgoing out-of-dialog requests via a
58 set of proxies by using a pre-loaded route-set defined by the Path headers in
59 the REGISTER request. See Realtime updates for more configuration information.
63 * The BRIDGE_FEATURES channel variable would previously only set features for
64 the calling party and would set this feature regardless of whether the
65 feature was in caps or in lowercase. Use of a caps feature for a letter
66 will now apply the feature to the calling party while use of a lowercase
67 letter will apply that feature to the called party.
69 * Add support for automixmonitor to the BRIDGE_FEATURES channel variable.
71 * PARKINGSLOT and PARKEDLOT channel variables will now be set for a parked
72 channel even when comebactoorigin=yes
76 * When performing queue pause/unpause on an interface without specifying an
77 individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
78 least one member of any queue exists for that interface.
80 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
81 for realtime queue log entries.
85 * Added the 'n' option to MeetMe to prevent application of the DENOISE function
86 to a channel joining a conference. Some channel drivers that vary the number
87 of audio samples in a voice frame will experience significant quality problems
88 if a denoiser is attached to the channel; this option gives them the ability
89 to remove the denoiser without having to unload func_speex.
93 * Add queue available hint. exten => 8501,hint,Queue:markq_avail
94 Note: the suffix '_avail' after the queuename.
95 Reports 'InUse' for no logged in agents or no free agents.
96 Reports 'Idle' when an agent is free.
100 * Redirecting reasons can now be set to arbitrary strings. This means
101 that the REDIRECTING dialplan function can be used to set the redirecting
102 reason to any string. It also allows for custom strings to be read as the
103 redirecting reason from SIP Diversion headers.
107 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
108 will store the path information for that peer when it registers. Realtime
109 tables can also use the 'supportpath' field to enable Path header support.
111 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
112 objectIdentifier. This maps to the supportpath option in sip.conf.
116 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
117 them, an Asterisk-specific version of pjproject needs to be installed.
118 Tarballs are available from https://github.com/asterisk/pjproject/tags/.
120 ------------------------------------------------------------------------------
121 --- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
122 ------------------------------------------------------------------------------
126 * The Asterisk build system will now build and install a shared library
127 (libasteriskssl.so) used to wrap various initialization and shutdown functions
128 from the libssl and libcrypto libraries provided by OpenSSL. This is done so
129 that Asterisk can ensure that these functions do *not* get called by any
130 modules that are loaded into Asterisk, since they should only be called once
131 in any single process. If desired, this feature can be disabled by supplying
132 the "--disable-asteriskssl" option to the configure script.
134 * A new make target, 'full', has been added to the Makefile. This performs
135 the same compilation actions as make all, but will also scan the entirety of
136 each source file for documentation. This option is needed to generate AMI
137 event documentation. Note that your system must have Python in order for
138 this make target to succeed.
140 * The optimization portion of the build system has been reworked to avoid
141 broken builds on certain architectures. All architecture-specific
142 optimization has been removed in favor of using -march=native to allow gcc
143 to detect the environment in which it is running when possible. This can
144 be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
146 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
147 make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
149 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
150 previously parsed the header file to obtain the version of Asterisk, you
151 will now have to go through Asterisk to get the version information.
159 * Added 'F()' option. Similar to the dial option, this can be supplied with
160 arguments indicating where the callee should go after the caller is hung up,
161 or without options specified, the priority after the Queue will be used.
166 * Added menu action admin_toggle_mute_participants. This will mute / unmute
167 all non-admin participants on a conference. The confbridge configuration
168 file also allows for the default sounds played to all conference users when
169 this occurs to be overriden using sound_participants_unmuted and
170 sound_participants_muted.
172 * Added menu action participant_count. This will playback the number of
173 current participants in a conference.
175 * Added announcement configuration option to user profile. If set the sound
176 file will be played to the user, and only the user, upon joining the
179 * Added record_file_append option that defaults to "yes", but if set to no
180 will create a new file between each start/stop recording.
185 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
186 channels respectively before the callee channels are called.
191 * Added support for IPv6.
193 * Add interrupt ('I') command to ExternalIVR. Sending this command from an
194 external process will cause the current playlist to be cleared, including
195 stopping any audio file that is currently playing. This is useful when you
196 want to interrupt audio playback only when specific DTMF is entered by the
202 * A new option, 'I' has been added to app_followme. By setting this option,
203 Asterisk will not update the caller with connected line changes when they
204 occur. This is similar to app_dial and app_queue.
206 * The 'N' option is now ignored if the call is already answered.
208 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
209 and caller channels respectively before the callee channels are called.
211 * The winning FollowMe outgoing call is now put on hold if the caller put it on
217 * MixMonitor hooks now have IDs associated with them which can be used to
218 assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
219 will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
220 now accepts that ID as an argument.
222 * Added 'm' option, which stores a copy of the recording as a voicemail in the
228 * The connect action in app_mysql now allows you to specify a port number to
229 connect to. This is useful if you run a MySQL server on a non-standard
235 * Increased the default number of allowed destinations from 5 to 12.
240 * The app_page application now no longer depends on DAHDI or app_meetme. It
241 has been re-architected to use app_confbridge internally.
246 * Added queue options autopausebusy and autopauseunavail for automatically
247 pausing a queue member when their device reports busy or congestion.
249 * The 'ignorebusy' option for queue members has been deprecated in favor of
250 the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
251 added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
252 per interface basis. Individual ringinuse values can now be set in
253 queues.conf via an argument to member definitions. Lastly, the queue
254 'ringinuse' setting now only determines defaults for the per member
255 'ringinuse' setting and does not override per member settings like it does
258 * Added 'F()' option. Similar to the dial option, this can be supplied with
259 arguments indicating where the callee should go after the caller is hung up,
260 or without options specified, the priority after the Queue will be used.
262 * Added new option log_member_name_as_agent, which will cause the membername to
263 be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
264 state_interface has been set.
266 * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
270 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
271 when receiving DTMF. Use the 'j' option to enable extension jumping. Also
272 changed arguments to SayUnixTime so that every option is truly optional even
273 when using multiple options (so that j option could be used without having to
274 manually specify timezone and format) There are other benefits, e.g., format
275 can now be used without specifying time zone as well.
280 * Addition of the VM_INFO function - see Function changes.
282 * The imapserver, imapport, and imapflags configuration options can now be
283 overriden on a user by user basis.
285 * When voicemail plays a message's envelope with saycid set to yes, when
286 reaching the caller id field it will play a recording of a file with the same
287 base name as the sender's callerid if there is a similarly named file in
288 <astspooldir>/recordings/callerids/
290 * Voicemails now contains a unique message identifier "msg_id", which is stored
291 in the message envelope with the sound files. IMAP backends will now store
292 the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
293 backends will store the message identifier in a "msg_id" column. See
294 UPGRADE.txt for more information.
296 * Added VoiceMailPlayMsg application. This application will play a single
297 voicemail message from a mailbox. The result of the application, SUCCESS or
298 FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
303 * Hangup handlers can be attached to channels using the CHANNEL() function.
304 Hangup handlers will run when the channel is hung up similar to the h
305 extension. The hangup_handler_push option will push a GoSub compatible
306 location in the dialplan onto the channel's hangup handler stack. The
307 hangup_handler_pop option will remove the last added location, and optionally
308 replace it with a new GoSub compatible location. The hangup_handler_wipe
309 option will remove all locations on the stack, and optionally add a new
312 * The expression parser now recognizes the ABS() absolute value function,
313 which will convert negative floating point values to positive values.
315 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
316 control of faxdetect.
318 * Addition of the VM_INFO function that can be used to retrieve voicemail
319 user information, such as the email address and full name.
320 The MAILBOX_EXISTS dialplan function has been deprecated in favour of
323 * The REDIRECTING function now supports the redirecting original party id
326 * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
327 lets you set some of the configuration options from the [general] section
328 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
329 the key sequence used to activate built-in features, such as blindxfer,
330 and automon. See the built-in documentation for details.
332 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
333 instead of simply the uri. This is the format that MessageSend() can use
334 in the from parameter for outgoing SIP messages.
336 * Added the PRESENCE_STATE function. This allows retrieving presence state
337 information from any presence state provider. It also allows setting
338 presence state information from a CustomPresence presence state provider.
339 See AMI/CLI changes for related commands.
341 * Added the AMI_CLIENT function to make manager account attributes available
342 to the dialplan. It currently supports returning the current number of
343 active sessions for a given account.
345 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
346 and the REDIRECTING functions.
354 * Added a manager event "LocalBridge" for local channel call bridges between
355 the two pseudo-channels created.
360 * Added dialtone_detect option for analog ports to disconnect incoming
361 calls when dialtone is detected.
363 * Added option colp_send to send ISDN connected line information. Allowed
364 settings are block, to not send any connected line information; connect, to
365 send connected line information on initial connect; and update, to send
366 information on any update during a call. Default is update.
368 * Add options namedcallgroup and namedpickupgroup to support installations
369 where a higher number of groups (>64) is required.
371 * Added support to use private party ID information with PRI calls.
376 * A new channel driver named chan_motif has been added which provides support for
377 Google Talk and Jingle in a single channel driver. This new channel driver includes
378 support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
379 hold, unhold, and ringing notification. It is also compliant with the current Jingle
380 specification, current Google Jingle specification, and the original Google Talk
386 * Added NAT support for RTP. Setting in config is 'nat', which can be set
387 globally and overriden on a peer by peer basis.
389 * Direct media functionality has been added. Options in config are:
390 directmedia (directrtp) and directrtpsetup (earlydirect)
392 * ChannelUpdate events now contain a CallRef header.
397 * Asterisk will no longer substitute CID number for CID name in the display
398 name field if CID number exists without a CID name. This change improves
399 compatibility with certain device features such as Avaya IP500's directory
402 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
403 created using that setting to not be removed during SIP reload.
405 * Added settings recordonfeature and recordofffeature. When receiving an INFO
406 request with a "Record:" header, this will turn the requested feature on/off.
407 Allowed values are 'automon', 'automixmon', and blank to disable. Note that
408 dynamic features must be enabled and configured properly on the requesting
409 channel for this to function properly.
411 * Add support to realtime for the 'callbackextension' option.
413 * When multiple peers exist with the same address, but differing
414 callbackextension options, incoming requests that are matched by address
415 will be matched to the peer with the matching callbackextension if it is
418 * Two new NAT options, auto_force_rport and auto_comedia, have been added
419 which set the force_rport and comedia options automatically if Asterisk
420 detects that an incoming SIP request crossed a NAT after being sent by
423 * The default global nat setting in sip.conf has been changed from force_rport
426 * NAT settings are now a combinable list of options. The equivalent of the
427 deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
429 * Adds an option send_diversion which can be disabled to prevent
430 diversion headers from automatically being added to INVITE requests.
432 * Add support for lightweight NAT keepalive. If enabled a blank packet will
433 be sent to the remote host at a given interval to keep the NAT mapping open.
434 This can be enabled using the keepalive configuration option.
436 * Add option 'tonezone' to specify country code for indications. This option
437 can be set both globally and overridden for specific peers.
439 * The SIP Security Events Framework now supports IPv6.
441 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
442 between multiple user agents. When set, for directmedia reinvites,
443 Asterisk will not send an immediate reinvite on an incoming call leg. This
444 option is useful when peered with another SIP user agent that is known to
445 send immediate direct media reinvites upon call establishment.
447 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
450 * Add options subminexpiry and submaxexpiry to set limits of subscription
451 timer independently from registration timer settings. The setting of the
452 registration timer limits still is done by options minexpiry, maxexpiry
453 and defaultexpiry. For backwards compatibility the setting of minexpiry
454 and maxexpiry also is used to configure the subscription timer limits if
455 subminexpiry and submaxexpiry are not set in sip.conf.
457 * Set registration timer limits to default values when reloading sip
458 configuration and values are not set by configuration.
460 * Add options namedcallgroup and namedpickupgroup to support installations
461 where a higher number of groups (>64) is required.
463 * When a MESSAGE request is received, the address the request was received from
464 is now saved in the SIP_RECVADDR variable.
466 * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
467 parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
468 the ANI2/OLI information is set on the channel, which can be retrieved using
469 the CALLERID function.
471 * Peers can now be configured to support negotiation of ICE candidates using
472 the setting icesupport. See res_rtp_asterisk changes for more information.
474 * Added support for format attribute negotiation. See the Codecs changes for
477 * Extra headers specified with SIPAddHeader are sent with the REFER message
478 when using Transfer application. See refer_addheaders in sip.conf.sample.
480 * Added support to use private party ID information with calls.
482 * Adds an option discard_remote_hold_retrieval that when set stops telling
483 the peer to start music on hold.
488 * Added skinny version 17 protocol support.
493 * Added ability to use multiple lines for a single phone. This allows multiple
494 calls to occur on a single phone, using callwaiting and switching between calls.
496 * Added option 'sharpdial' allowing end dialing by pressing # key
498 * Added option 'interdigit_timer' to control phone dial timeout
500 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
502 * Added global 'debug' option, that enables debug in channel driver
504 * Added ability to translate on-screen menu in multiple languages. Tested on
505 Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
506 ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
509 * In addition to English added French and Russian languages for on-screen menus
511 * Reworked dialing number input: added dialing by timeout, immediate dial on
512 on dialplan compare, phone number length now not limited by screen size
514 * Added ability to pickup a call using features.conf defined value and
520 * Add options namedcallgroup and namedpickupgroup to support installations
521 where a higher number of groups (>64) is required.
523 * Added support to use private party ID information with calls.
528 * The minimum DTMF duration can now be configured in asterisk.conf
529 as "mindtmfduration". The default value is (as before) set to 80 ms.
530 (previously it was only available in source code)
532 * Named ACLs can now be specified in acl.conf and used in configurations that
533 use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
534 used to specify an ACL, a similar form of 'acl' will add a named ACL to the
535 working ACL. In addition, some CLI commands have been added to provide
536 show information and allow for module reloading - see CLI Changes.
538 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
539 items (separated by commas), and items in the rule can be negated by prefixing
540 them with '!'. This simplifies Asterisk Realtime configurations, since it is no
541 longer necessray to control the order that the 'permit' and 'deny' columns are
542 returned from queries.
544 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
545 be used within the dynamic weight attribute when specifying a mapping.
547 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
548 header, instead of putting the user defined event name there. When enabled
549 the UserDefType header is added for user defined events. This feature is
550 enabled with the setting show_user_defined.
552 * Macro has been deprecated in favor of GoSub. For redirecting and connected
553 line purposes use the following variables instead of their macro equivalents:
554 REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
555 CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
556 cc_callback_macro in channel configurations.
558 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
561 * Call files now support the "early_media" option to connect with an outgoing
562 extension when early media is received.
564 * Added support to use private party ID information with calls.
569 * A new channel variable, AGIEXITONHANGUP, has been added which allows
570 Asterisk to behave like it did in Asterisk 1.4 and earlier where the
571 AGI application would exit immediately after a channel hangup is detected.
573 * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
574 are resolved and each address is attempted in turn until one succeeds or
578 AMI (Asterisk Manager Interface)
580 * The originate action now has an option "EarlyMedia" that enables the
581 call to bridge when we get early media in the call. Previously,
582 early media was disregarded always when originating calls using AMI.
584 * Added setvar= option to manager accounts (much like sip.conf)
586 * Originate now generates an error response if the extension given is not found
589 * MixMonitor will now show IDs associated with the mixmonitor upon creating
590 them if the i(variable) option is used. StopMixMonitor will accept
591 MixMonitorID as an option to close specific MixMonitors.
593 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
594 updated to include information about peers configured with
595 nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
596 detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
597 returned if auto_force_rport is not enabled.
599 * Added SIPpeerstatus manager command which will generate PeerStatus events
600 similar to the existing PeerStatus events found in chan_sip on demand.
602 * Hangup now can take a regular expression as the Channel option. If you want
603 to hangup multiple channels, use /regex/ as the Channel option. Existing
604 behavior to hanging up a single channel is unchanged, but if you pass a regex,
605 the manager will send you a list of channels back that were hung up.
607 * Support for IPv6 addresses has been added.
609 * AMI Events can now be documented in the Asterisk source. Note that AMI event
610 documentation is only generated when Asterisk is compiled using 'make full'.
611 See the CLI section for commands to display AMI event information.
613 * The AMI Hangup event now includes the AccountCode header so you can easily
614 correlate with AMI Newchannel events.
616 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
617 the StateInterface of the queue member.
619 * Added AMI event SessionTimeout in the Call category that is issued when a
620 call is terminated due to either RTP stream inactivity or SIP session timer
623 * CEL events can now contain a user defined header UserDefType. See core
624 changes for more information.
626 * OOH323 ChannelUpdate events now contain a CallRef header.
628 * Added PresenceState command. This command will report the presence state for
629 the given presence provider.
631 * Added Parkinglots command. This will list all parking lots as a series of
632 AMI Parkinglot events.
634 * Added MessageSend command. This behaves in the same manner as the
635 MessageSend application, and is a technolgoy agnostic mechanism to send out
636 of call text messages.
638 * Added "message" class authorization. This grants an account permission to
639 send out of call messages. Write-only.
644 * The "dialplan add include" command has been modified to create context a context
645 if one does not already exist. For instance, "dialplan add include foo into bar"
646 will create context "bar" if it does not already exist.
648 * A "dialplan remove context" command has been added to remove a context from
651 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
652 filenames of all running mixmonitors on a channel.
654 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
655 numeric instead of 0, 1, or 2.
657 * "stun show status" will show a table describing how the STUN client is
660 * "acl show [named acl]" will show information regarding a Named ACL. The
661 acl module can be reloaded with "reload acl".
663 * Added CLI command to display AMI event information - "manager show events",
664 which shows a list of all known and documented AMI events, and "manager show
665 event [event name]", which shows detail information about a specific AMI
668 * The result of the CLI command "queue show" now includes the state interface
669 information of the queue member.
671 * The command "core set verbose" will now set a separate level of logging for
672 each remote console without affecting any other console.
674 * Added command "cdr show pgsql status" to check connection status
676 * "sip show channel" will now display the complete route set.
678 * Added "presencestate list" command. This command will list all custom
679 presence states that have been set by using the PRESENCE_STATE dialplan
682 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
683 command. This changes a custom presence to a new state.
688 * Codec lists may now be modified by the '!' character, to allow succinct
689 specification of a list of codecs allowed and disallowed, without the
690 requirement to use two different keywords. For example, to specify all
691 codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
693 * Add support for parsing SDP attributes, generating SDP attributes, and
694 passing it through. This support includes codecs such as H.263, H.264, SILK,
695 and CELT. You are able to set up a call and have attribute information pass.
696 This should help considerably with video calls.
698 * The iLBC codec can now use a system-provided iLBC library if one is installed,
699 just like the GSM codec.
703 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
704 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
708 * Asterisk version and build information is now logged at the beginning of a
711 * Threads belonging to a particular call are now linked with callids which get
712 added to any log messages produced by those threads. Log messages can now be
713 easily identified as involved with a certain call by looking at their call id.
714 Call ids may also be attached to log messages for just about any case where
715 it can be determined to be related to a particular call.
717 * Each logging destination and console now have an independent notion of the
718 current verbosity level. Logger.conf now allows an optional argument to
719 the 'verbose' specifier, indicating the level of verbosity sent to that
720 particular logging destination. Additionally, remote consoles now each
721 have their own verbosity level. The command 'core set verbose' will now set
722 a separate level for each remote console without affecting any other
728 * Added 'announcement' option which will play at the start of MOH and between
729 songs in modes of MOH that can detect transitions between songs (eg.
735 * New per parking lot options: comebackcontext and comebackdialtime. See
736 configs/features.conf.sample for more details.
738 * Channel variable PARKER is now set when comebacktoorigin is disabled in
741 * Channel variable PARKEDCALL is now set with the name of the parking lot
742 when a timeout occurs.
748 CDR Postgresql Driver
750 * Added command "cdr show pgsql status" to check connection status
753 CDR Adaptive ODBC Driver
755 * Added schema option for databases that support specifying a schema.
763 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
764 CALENDAR_WRITE has completed successfully.
769 * A new option, 'probation' has been added to rtp.conf
770 RTP in strictrtp mode can now require more than 1 packet to exit learning
771 mode with a new source (and by default requires 4). The probation option
772 allows the user to change the required number of packets in sequence to any
773 desired value. Use a value of 1 to essentially restore the old behavior.
774 Also, with strictrtp on, Asterisk will now drop all packets until learning
775 mode has successfully exited. These changes are based on how pjmedia handles
776 media sources and source changes.
778 * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
779 enabled or disabled using the icesupport setting. A variety of other
780 settings have been introduced to configure STUN/TURN connections.
785 * A new module, res_corosync, has been introduced. This module uses the
786 Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
787 of Asterisk servers to both Message Waiting Indication (MWI) and/or
788 Device State (presence) information. This module is very similar to, and
789 is a replacement for the res_ais module that was in previous releases of
795 * This module adds a cleaned up, drop-in replacement for res_jabber called
796 res_xmpp. This provides the same externally facing functionality but is
797 implemented differently internally. res_jabber has been deprecated in favor
798 of res_xmpp; please see the UPGRADE.txt file for more information.
803 * The safe_asterisk script has been updated to allow several of its parameters
804 to be set from environment variables. This also enables a custom run
805 directory of Asterisk to be specified, instead of defaulting to /tmp.
807 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
808 its value to determine the directory to assume is the top-level directory of
809 the source tree. If the variable is not set, it defaults to the current
810 behavior and uses the current working directory.
812 ------------------------------------------------------------------------------
813 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
814 ------------------------------------------------------------------------------
818 * Asterisk now has protocol independent support for processing text messages
819 outside of a call. Messages are routed through the Asterisk dialplan.
820 SIP MESSAGE and XMPP are currently supported. There are options in
821 jabber.conf and sip.conf to allow enabling these features.
822 -> jabber.conf: see the "sendtodialplan" and "context" options.
823 -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
824 and "outofcall_message_context" options.
825 The MESSAGE() dialplan function and MessageSend() application have been
826 added to go along with this functionality. More detailed usage information
827 can be found on the Asterisk wiki (http://wiki.asterisk.org/).
828 * If real-time text support (T.140) is negotiated, it will be preferred for
829 sending text via the SendText application. For example, via SIP, messages
830 that were once sent via the SIP MESSAGE request would be sent via RTP if
831 T.140 text is negotiated for a call.
835 * parkedmusicclass can now be set for non-default parking lots.
837 Asterisk Manager Interface
838 --------------------------
839 * PeerStatus now includes Address and Port.
840 * Added Hold events for when the remote party puts the call on and off hold
841 for chan_dahdi ISDN channels.
842 * Added new action MeetmeListRooms to list active conferences (shows same
843 data as "meetme list" at the CLI).
844 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
845 Description field that is set by 'description' in the channel configuration
847 * Added Uniqueid header to UserEvent.
848 * Added new action FilterAdd to control event filters for the current session.
849 This requires the system permission and uses the same filter syntax as
850 filters that can be defined in manager.conf
851 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
852 versions had some instances of the event converted, but others were left
853 as-is. All Unlink events should now be converted to Bridge events. The AMI
854 protocol version number was incremented to 1.2 as a result of this change.
857 --------------------------
858 * The HTTP Server can bind to IPv6 addresses.
861 --------------------------
862 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
863 with busydetect. usage example: busypattern=200,200,200,600
866 --------------------------
867 * New 'gtalk show settings' command showing the current settings loaded from
869 * The 'logger reload' command now supports an optional argument, specifying an
870 alternate configuration file to use.
871 * 'dialplan add extension' command will now automatically create a context if
872 the specified context does not exist with a message indicated it did so.
873 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
874 Description field which can be populated with 'description' in the channel
875 configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
878 --------------------------
879 * The filter option in cdr_adaptive_odbc now supports negating the argument,
880 thus allowing records which do NOT match the specified filter.
881 * Added ability to log CONGESTION calls to CDR
884 --------------------------
885 * Ability to define custom SILK formats in codecs.conf.
886 * Addition of speex32 audio format with translation.
887 * CELT codec pass-through support and ability to define
888 custom CELT formats in codecs.conf.
889 * Ability to read raw signed linear files with sample rates
890 ranging from 8khz - 192khz. The new file extensions introduced
891 are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
892 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
893 Skinny, H.323, etc) can still only support the following codecs:
894 Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
895 siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
896 Video: h261, h263, h263p, h264, mpeg4
901 --------------------------
902 * New highly optimized and customizable ConfBridge application capable of
903 mixing audio at sample rates ranging from 8khz-96khz.
904 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
905 and bridge profiles on a channel.
906 * CONFBRIDGE_INFO dialplan function capable of retrieving information
907 about a conference such as locked status and number of parties, admins,
909 * Addition of video_mode option in confbridge.conf for adding video support
910 into a bridge profile.
911 * Addition of the follow_talker video_mode in confbridge.conf. This video
912 mode dynamically switches the video feed to always display the loudest talker
913 supplying video in the conference.
917 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
918 ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
919 variables from asterisk.conf.
923 * Addition of the JITTERBUFFER dialplan function. This function allows
924 for jitterbuffering to occur on the read side of a channel. By using
925 this function conference applications such as ConfBridge and MeetMe can
926 have the rx streams jitterbuffered before conference mixing occurs.
927 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
929 * Added STRREPLACE function. This function let's the user search a variable
930 for a given string to replace with another string as many times as the
931 user specifies or just throughout the whole string.
932 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
933 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
934 * Added extensions to chan_ooh323 in function CHANNEL()
936 libpri channel driver (chan_dahdi) DAHDI changes
937 --------------------------
938 * Added moh_signaling option to specify what to do when the channel's bridged
939 peer puts the ISDN channel on hold.
940 * Added display_send and display_receive options to control how the display ie
941 is handled. To send display text from the dialplan use the SendText()
942 application when the option is enabled.
943 * Added mcid_send option to allow sending a MCID request on a span.
946 --------------------------
947 * Added setvar option to calendar.conf to allow setting channel variables on
948 notification channels.
949 * Added "calendar show types" CLI command to list registered calendar
953 --------------------------
954 * Added two new options, r and t with file name arguments to record
955 single direction (unmixed) audio recording separate from the bidirectional
956 (mixed) recording. The mixed file name argument is optional now as long
957 as at least one recording option is used.
960 --------------------------
961 * Added a new option, l, which will disable local call optimization for
962 channels involved with the FollowMe thread. Use this option to improve
963 compatability for a FollowMe call with certain dialplan apps, options, and
967 --------------------------
968 * Added option "k" that will automatically close the conference when there's
969 only one person left when a user exits the conference.
972 --------------------------
973 * cel_pgsql now supports the 'extra' column for data added using the
974 CELGenUserEvent() application.
977 --------------------------
978 * Support for defining hints has been added to pbx_lua. See the 'hints' table
979 in the sample extensions.lua file for syntax details.
980 * Applications that perform jumps in the dialplan such as Goto will now
981 execute properly. When pbx_lua detects that the context, extension, or
982 priority we are executing on has changed it will immediately return control
983 to the asterisk PBX engine. Currently the engine cannot detect a Goto to
984 the priority after the currently executing priority.
985 * An autoservice is now started by default for pbx_lua channels. It can be
986 stopped and restarted using the autoservice_stop() and autoservice_start()
990 --------------------------
991 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
992 into a FAXStatus event with an 'Operation' header that will be either
993 'send', 'receive', and 'gateway'.
994 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
995 Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
996 feature will handle converting a fax call between an audio T.30 fax terminal
997 and an IFP T.38 fax terminal.
1001 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
1002 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
1003 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
1007 * Added general option negative_penalty_invalid default off. when set
1008 members are seen as invalid/logged out when there penalty is negative.
1009 for realtime members when set remove from queue will set penalty to -1.
1010 * Added queue option autopausedelay when autopause is enabled it will be
1011 delayed for this number of seconds since last successful call if there
1012 was no prior call the agent will be autopaused immediately.
1013 * Added member option ignorebusy this when set and ringinuse is not
1014 will allow per member control of multiple calls as ringinuse does for
1019 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
1021 * Added 'k' option to MeetMe to automatically kill the conference when there's only
1022 one participant left (much like a normal call bridge)
1023 * Added extra argument to Originate to set timeout.
1027 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
1028 SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
1029 utility in the UTILS section of menuselect. If an existing astdb is found and no
1030 astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
1031 convert an existing astdb to the SQLite3 version automatically at runtime.
1035 * Modules marked as deprecated are no longer marked as building by default. Enabling
1036 these modules is still available via menuselect.
1040 * authdebug is now disabled by default. To enable this functionaility again
1041 set authdebug = yes in iax.conf.
1045 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
1046 releases it was disabled.
1050 * The PBX core previously made a call with a non-existing extension test for
1051 extension s@default and jump there if the extension existed.
1052 This was a bad default behaviour and violated the principle of least surprise.
1053 It has therefore been changed in this release. It may affect some
1054 applications and configurations that rely on this behaviour. Most channel
1055 drivers have avoided this for many releases by testing whether the extension
1056 called exists before starting the PBX and generating a local error.
1057 This behaviour still exists and works as before.
1059 Extension "s" is used when no extension is given in a channel driver,
1060 like immediate answer in DAHDI or calling to a domain with no user part
1063 ------------------------------------------------------------------------------
1064 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
1065 ------------------------------------------------------------------------------
1069 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
1070 now defaults to force_rport. It is very important that phones requiring nat=no be
1071 specifically set as such instead of relying on the default setting. If at all
1072 possible, all devices should have nat settings configured in the general section as
1073 opposed to configuring nat per-device.
1074 * Added preferred_codec_only option in sip.conf. This feature limits the joint
1075 codecs sent in response to an INVITE to the single most preferred codec.
1076 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
1077 to be used for the outgoing call. It must be one of the codecs configured
1079 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
1080 to be used for holding a private key. If tlsprivatekey is not specified,
1081 tlscertfile is searched for both public and private key.
1082 * Added tlsclientmethod option to sip.conf. This allows the protocol for
1083 outbound client connections to be specified.
1084 * The sendrpid parameter has been expanded to include the options
1085 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
1086 header to be sent (equivalent to setting sendrpid=yes) and setting
1087 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
1088 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
1089 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
1090 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
1091 will accept the SDP even if the SDP version number is not properly incremented,
1092 but will generate a warning in the log indicating that the SIP peer that sent
1093 the SDP should have the 'ignoresdpversion' option set.
1094 * The 'nat' option has now been been changed to have yes, no, force_rport, and
1095 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
1096 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
1097 remote side requests it and disables symmetric RTP support. Setting it to
1098 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
1099 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
1100 and enables symmetric RTP support.
1101 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
1102 response. This permits the master channel to know how each channel dialled
1103 in a multi-channel setup resolved in an individual way. This carries a
1104 performance penalty and can be disabled in sip.conf using the
1105 'storesipcause' option.
1106 * Added 'externtcpport' and 'externtlsport' options to allow custom port
1107 configuration for the externip and externhost options when tcp or tls is used.
1108 * Added support for message body (stored in content variable) to SIP NOTIFY message
1109 accessible via AMI and CLI.
1110 * Added 'media_address' configuration option which can be used to explicitly specify
1111 the IP address to use in the SDP for media (audio, video, and text) streams.
1112 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
1113 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
1115 * Added 'use_q850_reason' configuration option for generating and parsing
1116 if available Reason: Q.850;cause=<cause code> header. It is implemented
1117 in some gateways for better passing PRI/SS7 cause codes via SIP.
1118 * When dialing SIP peers, a new component may be added to the end of the dialstring
1119 to indicate that a specific remote IP address or host should be used when dialing
1120 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
1121 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
1122 ability to selectively force bridged channels to also be encrypted is also
1123 implemented. Branching in the dialplan can be done based on whether or not
1124 a channel has secure media and/or signaling.
1125 * Added directmediapermit/directmediadeny to limit which peers can send direct media
1127 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
1128 Charge messages to snom phones.
1129 * Added support for G.719 media streams.
1130 * Added support for 16khz signed linear media streams.
1131 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
1132 RTP has been outfitted with the same abilities.
1133 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
1134 available in device configurations as well as in the dial plan.
1135 * Addition of the 'subscribe_network_change' option for turning on and off
1136 res_stun_monitor module support in chan_sip.
1137 * Addition of the 'auth_options_requests' option for turning on and off
1138 authentication for OPTIONS requests in chan_sip.
1142 * Add #tryinclude statement for config files. This provides the same
1143 functionality as the #include statement however an asterisk module will
1144 still load if the filename does not exist. Using the #include statement
1145 Asterisk will not allow the module to load.
1149 * Added rtsavesysname option into iax.conf to allow the systname to be saved
1150 on realtime updates.
1151 * Added the ability for chan_iax2 to inform the dialplan whether or not
1152 encryption is being used. This interoperates with the SIP SRTP implementation
1153 so that a secure SIP call can be bridged to a secure IAX call when the
1154 dialplan requires bridged channels to be "secure".
1155 * Addition of the 'subscribe_network_change' option for turning on and off
1156 res_stun_monitor module support in chan_iax.
1161 * Added ability to preset channel variables on indicated lines with the setvar
1162 configuration option. Also, clearvars=all resets the list of variables back
1164 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
1165 See configs/res_pktccops.conf for more information.
1167 XMPP Google Talk/Jingle changes
1168 -------------------------------
1169 * Added the externip option to gtalk.conf.
1170 * Added the stunaddr option to gtalk.conf which allows for the automatic
1171 retrieval of the external ip from a stun server.
1175 * Added 'p' option to PickupChan() to allow for picking up channel by the first
1176 match to a partial channel name.
1177 * Added .m3u support for Mp3Player application.
1178 * Added progress option to the app_dial D() option. When progress DTMF is
1179 present, those values are sent immediately upon receiving a PROGRESS message
1180 regardless if the call has been answered or not.
1181 * Added functionality to the app_dial F() option to continue with execution
1182 at the current location when no parameters are provided.
1183 * Added the 'a' option to app_dial to answer the calling channel before any
1184 announcements or macros are executed.
1185 * Modified app_dial to set answertime when the called channel answers even if
1186 the called channel hangs up during playback of an announcement.
1187 * Modified app_dial 'r' option to support an additional parameter to play an
1188 indication tone from indications.conf
1189 * Added c() option to app_chanspy. This option allows custom DTMF to be set
1190 to cycle through the next available channel. By default this is still '*'.
1191 * Added x() option to app_chanspy. This option allows DTMF to be set to
1192 exit the application.
1193 * The Voicemail application has been improved to automatically ignore messages
1194 that only contain silence.
1195 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
1196 associated mailbox(es) to be greetings-only.
1197 * The ChanSpy application now has the 'S' option, which makes the application
1198 automatically exit once it hits a point where no more channels are available
1200 * The ChanSpy application also now has the 'E' option, which spies on a single
1201 channel and exits when that channel hangs up.
1202 * The MeetMe application now turns on the DENOISE() function by default, for
1203 each participant. In our tests, this has significantly decreased background
1204 noise (especially noisy data centers).
1205 * Voicemail now permits storage of secrets in a separate file, located in the
1206 spool directory of each individual user. The control for this is located in
1207 the "passwordlocation" option in voicemail.conf. Please see the sample
1208 configuration for more information.
1209 * The ChanIsAvail application now exposes the returned cause code using a separate
1210 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
1211 * Added 'd' option to app_followme. This option disables the "Please hold"
1213 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
1214 received will terminate recording.
1215 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
1216 Previously the folder could only be set per context, but has now been extended
1217 using the imapfolder option.
1218 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
1219 * Voicemail now allows the pager date format to be specified separately from the
1221 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
1222 to allow joining, leaving, and sending text to group chats.
1223 * MeetMe has a new option 'G' to play an announcement before joining a conference.
1224 * Page has a new option 'A(x)' which will playback an announcement simultaneously
1225 to all paged phones (and optionally excluding the caller's one using the new
1226 option 'n') before the call is bridged.
1227 * The 'f' option to Dial has been augmented to take an optional argument. If no
1228 argument is provided, the 'f' option works as it always has. If an argument is
1229 provided, then the connected party information of all outgoing channels created
1230 during the Dial will be set to the argument passed to the 'f' option.
1231 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
1233 * The OSP lookup application adds in/outbound network ID, optional security,
1234 number portability, QoS reporting, destination IP port, custom info and service
1236 * Added new application VMSayName that will play the recorded name of the voicemail
1237 user if it exists, otherwise will play the mailbox number.
1238 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
1239 retrieve state for a particular bridge, where <name> is the conference name
1240 * app_directory now allows exiting at any time using the operator or pound key.
1241 * Voicemail now supports setting a locale per-mailbox.
1242 * Two new applications are provided for declining counting phrases in multiple
1243 languages. See the application notes for SayCountedNoun and SayCountedAdj for
1245 * Voicemail now runs the externnotify script when pollmailboxes is activated and
1247 * Voicemail now includes rdnis within msgXXXX.txt file.
1248 * ExternalIVR now supports IPv6 addresses.
1249 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
1250 at https://wiki.asterisk.org/wiki/x/oQBB
1251 * ParkedCall and Park can now specify the parking lot to use.
1255 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
1256 over SRV records associated with a specific service. From the CLI, type
1257 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
1258 details on how these may be used.
1259 * PITCH_SHIFT dialplan function added. This function can be used to modify the
1260 pitch of a channel's tx and rx audio streams.
1261 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
1262 setting various connected line and redirecting party information.
1263 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
1264 support ISDN subaddressing.
1265 * The CHANNEL() function now supports the "name" and "checkhangup" options.
1266 * For DAHDI channels, the CHANNEL() dialplan function now allows
1267 the dialplan to request changes in the configuration of the active
1268 echo canceller on the channel (if any), for the current call only.
1271 exten => s,n,Set(CHANNEL(echocan_mode)=off)
1273 The possible values are:
1275 on - normal mode (the echo canceller is actually reinitialized)
1277 fax - FAX/data mode (NLP disabled if possible, otherwise completely
1279 voice - voice mode (returns from FAX mode, reverting the changes that
1280 were made when FAX mode was requested)
1281 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
1282 and setting variables on the channel which created the current channel.
1283 Administrators should take care to avoid naming conflicts, when multiple
1284 channels are dialled at once, especially when used with the Local channel
1285 construct (which all could set variables on the master channel). Usage
1286 of the HASH() dialplan function, with the key set to the name of the slave
1287 channel, is one approach that will avoid conflicts.
1288 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
1290 * func_odbc now allows multiple row results to be retrieved without using
1291 mode=multirow. If rowlimit is set, then additional rows may be retrieved
1292 from the same query by using the name of the function which retrieved the
1293 first row as an argument to ODBC_FETCH().
1294 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
1295 dialplan. This function returns the content of the received message.
1296 * Added REPLACE, which searches a given variable name for a set of characters,
1297 then either replaces them with a single character or deletes them.
1298 * Added PASSTHRU, which literally passes the same argument back as its return
1299 value. The intent is to be able to use a literal string argument to
1300 functions that currently require a variable name as an argument.
1301 * HASH-associated variables now can be inherited across channel creation, by
1302 prefixing the name of the hash at assignment with the appropriate number of
1303 underscores, just like variables.
1304 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
1305 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
1306 whether or not channels that are bridged to the current channel will be
1307 required to have secure signaling and/or media.
1308 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
1309 the current channel has secure signaling and/or media.
1310 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
1311 "no_media_path" option.
1312 Returns "0" if there is a B channel associated with the call.
1313 Returns "1" if no B channel is associated with the call. The call is either
1314 on hold or is a call waiting call.
1315 * Added option to dialplan function CDR(), the 'f' option
1316 allows for high resolution times for billsec and duration fields.
1317 * FILE() now supports line-mode and writing.
1318 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
1319 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
1323 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
1324 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
1325 and is set when a dynamic feature is triggered.
1326 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
1327 to dynamically create a new parking lot matching the value this varible is
1329 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
1330 features.conf that should be the base for dynamic parkinglots.
1331 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
1332 parkinglot should have.
1333 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
1334 parkinglot should have.
1335 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
1340 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
1341 timeout has expired.
1342 * Added 'R' option to app_queue. This option stops moh and indicates ringing
1343 to the caller when an Agent's phone is ringing. This can be used to indicate
1344 to the caller that their call is about to be picked up, which is nice when
1345 one has been on hold for an extened period of time.
1346 * A new config option, penaltymemberslimit, has been added to queues.conf.
1347 When set this option will disregard penalty settings when a queue has too
1349 * A new option, 'I' has been added to both app_queue and app_dial.
1350 By setting this option, Asterisk will not update the caller with
1351 connected line changes or redirecting party changes when they occur.
1352 * A 'relative-periodic-announce' option has been added to queues.conf. When
1353 enabled, this option will cause periodic announce times to be calculated
1354 from the end of announcements rather than from the beginning.
1355 * The autopause option in queues.conf can be passed a new value, "all." The
1356 result is that if a member becomes auto-paused, he will be paused in all
1357 queues for which he is a member, not just the queue that failed to reach
1359 * Added dialplan function QUEUE_EXISTS to check if a queue exists
1360 * The queue logger now allows events to optionally propagate to a file,
1361 even when realtime logging is turned on. Additionally, realtime logging
1362 supports sending the event arguments to 5 individual fields, although it
1363 will fallback to the previous data definition, if the new table layout is
1366 mISDN channel driver (chan_misdn) changes
1367 ----------------------------------------
1368 * Added display_connected parameter to misdn.conf to put a display string
1369 in the CONNECT message containing the connected name and/or number if
1370 the presentation setting permits it.
1371 * Added display_setup parameter to misdn.conf to put a display string
1372 in the SETUP message containing the caller name and/or number if the
1373 presentation setting permits it.
1374 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
1375 indicate the dialplan settings are to be obtained from the asterisk
1377 * Made misdn.conf parameter callerid accept the "name" <number> format
1378 used by the rest of the system.
1379 * Made use the nationalprefix and internationalprefix misdn.conf
1380 parameters to prefix any received number from the ISDN link if that
1381 number has the corresponding Type-Of-Number. NOTE: This includes
1382 comparing the incoming call's dialed number against the MSN list.
1383 * Added the following new parameters: unknownprefix, netspecificprefix,
1384 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
1385 received number from the ISDN link if that number has the corresponding
1387 * Added new dialplan application misdn_command which permits controlling
1388 the CCBS/CCNR functionality.
1389 * Added new dialplan function mISDN_CC which permits retrieval of various
1390 values from an active call completion record.
1391 * For PTP, you should manually send the COLR of the redirected-to party
1392 for an incomming redirected call if the incoming call could experience
1393 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
1394 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
1395 if the REDIRECTING(from-num) is not empty.
1396 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
1397 option on all of the REDIRECTING statements before dialing the
1398 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
1399 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
1400 redirecting-to presentation (COLR) when it becomes available.
1401 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
1404 thirdparty mISDN enhancements
1405 -----------------------------
1406 mISDN has been modified by Digium, Inc. to greatly expand facility message
1408 * Enhanced COLP support for call diversion and transfer.
1409 * CCBS/CCNR support.
1411 The latest modified mISDN v1.1.x based version is available at:
1412 http://svn.digium.com/svn/thirdparty/mISDN/trunk
1413 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
1415 Tagged versions of the modified mISDN code are available under:
1416 http://svn.digium.com/svn/thirdparty/mISDN/tags
1417 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
1419 libpri channel driver (chan_dahdi) DAHDI changes
1420 -------------------------------------------
1421 * The channel variable PRIREDIRECTREASON is now just a status variable
1422 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
1423 to read and alter the reason.
1424 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
1425 redirected-to party for an incomming redirected call if the incoming call
1426 could experience further redirects. Just set the
1427 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
1428 to the COLR. A call has been redirected if the REDIRECTING(count) is not
1430 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
1431 use the inhibit(i) option on all of the REDIRECTING statements before
1432 dialing the redirected-to party. You still have to set the
1433 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
1434 will update the redirecting-to presentation (COLR) when it becomes available.
1435 * Added the ability to ignore calls that are not in a Multiple Subscriber
1436 Number (MSN) list for PTMP CPE interfaces.
1437 * Added dynamic range compression support for dahdi channels. It is
1438 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
1439 * Added support for ISDN calling and called subaddress with partial support
1440 for connected line subaddress.
1441 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
1442 * Added handling of received HOLD/RETRIEVE messages and the optional ability
1443 to transfer a held call on disconnect similar to an analog phone.
1444 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
1445 Will reroute/deflect an outgoing call when receive the message.
1446 Can use the DAHDISendCallreroutingFacility to send the message for the
1448 * Added standard location to add options to chan_dahdi dialing:
1449 Dial(DAHDI/g1[/extension[/options]])
1452 R Reverse charging indication
1453 * Added Reverse Charging Indication (Collect calls) send/receive option.
1454 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
1455 Dial(DAHDI/g1/extension/R)
1456 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
1457 (requires latest LibPRI)
1458 * Added ability to send/receive keypad digits in the SETUP message.
1459 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
1460 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
1461 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
1462 (requires latest LibPRI)
1463 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
1464 to eliminate tromboned calls. A tromboned call goes out an interface and comes
1465 back into the same interface. Tromboned calls happen because of call routing,
1466 call deflection, call forwarding, and call transfer.
1467 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
1468 * Added the ability to support call waiting calls. (The SETUP has no B channel
1470 * Added Malicious Call ID (MCID) event to the AMI call event class.
1471 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
1473 Asterisk Manager Interface
1474 --------------------------
1475 * The Hangup action now accepts a Cause header which may be used to
1476 set the channel's hangup cause.
1477 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
1478 to specify a separate .pem file to hold a private key. By default sslcert
1479 is used to hold both the public and private key.
1480 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
1481 for options containing the 'tls' prefix. For example, 'sslenable' is now
1482 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
1483 across all .conf files. All affected sample.conf files have been modified to
1484 reflect this change. Previous options such as 'sslenable' still work,
1485 but options with the 'tls' prefix are preferred.
1486 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
1487 in a channel. (res_mutestream.so)
1488 * The configuration file manager.conf now supports a channelvars option, which
1489 specifies a list of channel variables to include in each channel-oriented
1491 * The redirect command now has new parameters ExtraContext, ExtraExtension,
1492 and ExtraPriority to allow redirecting the second channel to a different
1493 location than the first.
1494 * Added new event "JabberStatus" in the Jabber module to monitor buddies
1496 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
1497 in a MixMonitor recording.
1498 * The 'iax2 show peers' output is now similar to the expected output of
1500 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
1502 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
1503 AOC-E messages on a channel.
1504 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
1505 conform more closely to similar events.
1506 * Added a new eventfilter option per user to allow whitelisting and blacklisting
1508 * Added optional parkinglot variable for park command.
1509 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
1510 if CallerIDNum and CallerIDName headers are also present.
1512 Channel Event Logging
1513 ---------------------
1514 * A new interface, CEL, is introduced here. CEL logs single events, much like
1515 the AMI, but it differs from the AMI in that it logs to db backends much
1516 like CDR does; is based on the event subsystem introduced by Russell, and
1517 can share in all its benefits; allows multiple backends to operate like CDR;
1518 is specialized to event data that would be of concern to billing sytems,
1519 like CDR. Backends for logging and accounting calls have been produced,
1520 but a new CDR backend is still in development.
1524 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
1525 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
1526 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
1527 * Multiple files and formats can now be specified in cdr_custom.conf.
1528 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
1529 See configs/cdr_syslog.conf.sample for more information.
1530 * A 'sequence' field has been added to CDRs which can be combined with
1531 linkedid or uniqueid to uniquely identify a CDR.
1532 * Handling of billsec and duration field has changed. If your table definition
1533 specifies those fields as float,double or similar they will now be logged with
1534 microsecond accuracy instead of a whole integer.
1536 Calendaring for Asterisk
1537 ------------------------
1538 * A new set of modules were added supporing calendar integration with Asterisk.
1539 Dialplan functions for reading from and writing to calendars are included,
1540 as well as the ability to execute dialplan logic upon calendar event notifications.
1541 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
1542 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
1543 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
1544 2003 support does not support forms-based authentication).
1546 Call Completion Supplementary Services for Asterisk
1547 ---------------------------------------------------
1548 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
1549 DAHDI/ISDN supports call completion for the following switch types:
1550 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
1551 See https://wiki.asterisk.org/wiki/x/2ABQ for details.
1553 Multicast RTP Support
1554 ---------------------
1555 * A new RTP engine and channel driver have been added which supports Multicast RTP.
1556 The channel driver can be used with the Page application to perform multicast RTP
1557 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
1558 Type can be either basic or linksys.
1559 Destination is the IP address and port for the RTP packets.
1560 Control address is specific to the linksys type and is used for sending the control
1561 packets unique to them.
1563 Security Events Framework
1564 -------------------------
1565 * Asterisk has a new C API for reporting security events. The module res_security_log
1566 sends these events to the "security" logger level. Currently, AMI is the only
1567 Asterisk component that reports security events. However, SIP support will be
1568 coming soon. For more information on the security events framework, see the
1569 "Asterisk Security Framework" section of the Asterisk wiki at
1570 https://wiki.asterisk.org/wiki/x/wgBQ
1571 * SIP support was added in Asterisk 10
1572 * This API now supports IPv6 addresses
1576 * A technology independent fax frontend (res_fax) has been added to Asterisk.
1577 * A spandsp based fax backend (res_fax_spandsp) has been added.
1578 * The app_fax module has been deprecated in favor of the res_fax module and
1579 the new res_fax_spandsp backend.
1580 * The SendFAX and ReceiveFAX applications now send their log messages to a
1581 'fax' logger level, instead of to the generic logger levels. To see these
1582 messages, the system's logger.conf file will need to direct the 'fax' logger
1583 level to one or more destinations; the logger.conf.sample file includes an
1584 example of how to do this. Note that if the 'fax' logger level is *not*
1585 directed to at least one destination, log messages generated by these
1586 applications will be lost, and that if the 'fax' logger level is directed to
1587 the console, the 'core set verbose' and 'core set debug' CLI commands will
1588 have no effect on whether the messages appear on the console or not.
1592 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
1593 Now, in order to enable transmitting silence during record the transmit_silence
1594 option should be used. transmit_silence_during_record remains a valid option, but
1595 defaults to the behavior of the transmit_silence option.
1596 * Addition of the Unit Test Framework API for managing registration and execution
1597 of unit tests with the purpose of verifying the operation of C functions.
1598 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
1599 XMPP text messages to the remote JID.
1600 * Modules.conf has a new option - "require" - that marks a module as critical for
1601 the execution of Asterisk.
1602 If one of the required modules fail to load, Asterisk will exit with a return
1604 * An 'X' option has been added to the asterisk application which enables #exec support.
1605 This allows #exec to be used in asterisk.conf.
1606 * jabber.conf supports a new option auth_policy that toggles auto user registration.
1607 * A new lockconfdir option has been added to asterisk.conf to protect the
1608 configuration directory (/etc/asterisk by default) during reloads.
1609 * The parkeddynamic option has been added to features.conf to enable the creation
1610 of dynamic parkinglots.
1611 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
1612 the reportalarms config option.
1613 * chan_dahdi supports dialing configuring and dialing by device file name.
1614 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
1615 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
1616 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
1617 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
1618 Handy for the above name-based syntax as it does not depend on
1619 initialization order.
1620 * The Realtime dialplan switch now caches entries for 1 second. This provides a
1621 significant increase in performance (about 3X) for installations using this switchtype.
1622 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
1623 AIS. For more information, please see the Distributed Device State section of the
1624 Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1625 * The addition of G.719 pass-through support.
1626 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
1627 during device configuration.
1628 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
1629 have less than 3 lines on the LCD.
1630 * Realtime now supports database failover. See the sample extconfig.conf for details.
1631 * The addition of improved translation path building for wideband codecs. Sample
1632 rate changes during translation are now avoided unless absolutely necessary.
1633 * The addition of the res_stun_monitor module for monitoring and reacting to network
1634 changes while behind a NAT.
1635 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
1636 DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
1637 These allow support for any Administration. Default is AT&T values.
1641 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
1642 optionally accept a filename, to apply the setting only to the code generated from
1643 that source file when Asterisk was built. However, there are some modules in Asterisk
1644 that are composed of multiple source files, so this did not result in the behavior
1645 that users expected. In this version, 'core set debug' and 'core set verbose'
1646 can optionally accept *module* names instead (with or without the .so extension),
1647 which applies the setting to the entire module specified, regardless of which source
1648 files it was built from.
1649 * New 'manager show settings' command showing the current settings loaded from
1651 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
1652 the channel hangup request to all channels.
1653 * Added a "core reload" CLI command that executes a global reload of Asterisk.
1655 ------------------------------------------------------------------------------
1656 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
1657 ------------------------------------------------------------------------------
1661 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
1662 Snom phones use this for call pickup of extensions that the phone is
1664 * Added support for setting the domain in the URI for caller of an
1665 outbound call by using the SIPFROMDOMAIN channel variable.
1666 * Added a new configuration option "remotesecret" for authentication to
1667 remote services. For backwards compatibility, "secret" still has the
1668 same function as before, but now you can configure both a remote secret and a
1669 local secret for mutual authentication.
1670 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
1671 the sound will be played to the target of an attended transfer
1672 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
1673 finer control over how many peers Asterisk will qualify and the gap between them
1674 when all peers need to be qualified at the same time.
1675 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
1676 (either globally or for a specific peer), chan_sip will treat any SDP data
1677 it receives as new data and update the media stream accordingly. By
1678 default, Asterisk will only modify the media stream if the SDP session
1679 version received is different from the current SDP session version. This
1680 option is required to interoperate with devices that have non-standard SDP
1681 session version implementations (observed with Microsoft OCS). This option
1682 is disabled by default.
1683 * The parsing of register => lines in sip.conf has been modified to allow a port
1684 to be present in the "user" portion. Please see the sip.conf.sample file for more
1686 * Added support for subscribing to MWI on a remote server and making the status available
1687 as a mailbox. Please see the sip.conf.sample file for more information.
1688 * Added a function to remove SIP headers added in the dialplan before the
1689 first INVITE is generated - SIPRemoveHeader()
1690 * Channel variables set with setvar= in a device configuration is now
1691 set both for inbound and outbound calls.
1692 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
1696 * Added immediate option to iax.conf
1697 * Added forceencryption option to iax.conf
1698 * Added Encryption and Trunk status to manager command "iaxpeers"
1702 * The configuration file now holds separate sections for devices and lines.
1703 Please have a look at configs/skinny.conf.sample and change your skinny.conf
1708 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
1709 support for LibOpenR2. http://www.libopenr2.org/
1710 * The UK option waitfordialtone has been added for use with BT analog
1712 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
1713 is used in conjunction with the 'faxdetect' configuration option. When
1714 'faxbuffers' is used and fax tones are detected, the channel will dynamically
1715 switch to the configured faxbuffers policy. For example, to use 6 buffers
1716 and a 'full' buffer policy for a fax transmission, add:
1718 The faxbuffers configuration will be in affect until the call is torn down.
1719 * Added service message support for 4ESS/5ESS switches.
1723 * For DAHDI channels, the CHANNEL() dialplan function now
1724 supports changing the channel's buffer policy (for the current
1725 call only), using this syntax:
1727 exten => s,n,Set(CHANNEL(buffers)=6,full)
1729 This would change the channel to the 'full' buffer policy and
1730 6 (six) buffers. Possible options for this setting are the same
1731 as those in chan_dahdi.conf.
1732 * Added a new dialplan function, CURLOPT, which permits setting various
1733 options that may be useful with the CURL dialplan function, such as
1734 cookies, proxies, connection timeouts, passwords, etc.
1735 * Permit the syntax and synopsis fields of the corresponding dialplan
1736 functions to be individually set from func_odbc.conf.
1737 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
1738 * func_odbc now may specify an insert query to execute, when the write query
1739 affects 0 rows (usually indicating that no such row exists).
1740 * Added a new dialplan function, LISTFILTER, which permits removing elements
1741 from a set list, by name. Uses the same general syntax as the existing CUT
1742 and FIELDQTY dialplan functions, which also manage lists.
1743 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
1744 obtaining realtime data from the dialplan.
1745 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
1746 a subroutine when using the GoSub() and Return() applications.
1747 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
1748 of "core show function AUDIOHOOK_INHERIT" from the CLI
1749 * Added AES_ENCRYPT. For information on its use, please see the output
1750 of "core show function AES_ENCRYPT" from the CLI
1751 * Added AES_DECRYPT. For information on its use, please see the output
1752 of "core show function AES_DECRYPT" from the CLI
1753 * func_odbc now supports database transactions across multiple queries.
1757 * Scheduled meetme conferences may now have their end times extended by
1759 * app_authenticate now gives the ability to select a prompt other than
1761 * app_directory now pays attention to the searchcontexts setting in
1762 voicemail.conf and will look through all contexts, if no context is
1763 specified in the initial argument.
1764 * A new application, Originate, has been introduced, that allows asynchronous
1765 call origination from the dialplan.
1766 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
1767 in addition to the setting in the "general" context.
1768 * Added ConfBridge dialplan application which does conference bridges without
1769 DAHDI. For information on its use, please see the output of
1770 "core show application ConfBridge" from the CLI.
1774 * The Asterisk CLI has a new command, "channel redirect", which is similar in
1775 operation to the AMI Redirect action.
1776 * extensions.conf now allows you to use keyword "same" to define an extension
1777 without actually specifying an extension. It uses exactly the same pattern
1778 as previously used on the last "exten" line. For example:
1779 exten => 123,1,NoOp(something)
1780 same => n,SomethingElse()
1781 * musiconhold.conf classes of type 'files' can now use relative directory paths,
1782 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
1783 * All deprecated CLI commands are removed from the sourcecode. They are now handled
1784 by the new clialiases module. See cli_aliases.conf.sample file.
1785 * Times within timespecs are now accurate down to the minute. This is a change
1786 from historical Asterisk, which only provided timespecs rounded to the nearest
1787 even (read: evenly divisible by 2) minute mark.
1788 * The realtime switch now supports an option flag, 'p', which disables searches for
1790 * In addition to a time range and date range, timespecs now accept a 5th optional
1791 argument, timezone. This allows you to perform time checks on alternate
1792 timezones, especially if those daylight savings time ranges vary from your
1793 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
1795 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
1796 give you the correct output for an asterisk box behind nat. It will give you the
1797 externhost and localnet settings.
1798 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
1799 can connect calls in passthrough mode, as well as record and play back files.
1800 * Successful and unsuccessful call pickup can now be alerted through sounds, by
1801 using pickupsound and pickupfailsound in features.conf.
1802 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
1803 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
1804 instead of the /var/run/asterisk.pid where it used to be. This will make
1805 installs as non-root easier to manage.
1810 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
1811 be written; they will no longer be explicitly written.
1813 Asterisk Manager Interface
1814 --------------------------
1815 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
1816 a non-empty value) in your request. If you do this, any pending AMI events will
1817 *not* be included in the response to your request as they would normally, but
1818 will be left in the event queue for the next request you make to retrieve. For
1819 some applications, this will allow you to guarantee that you will only see
1820 events in responses to 'WaitEvent' actions, and can better know when to expect them.
1821 To know whether the Asterisk server supports this header or not, your client can
1822 inspect the first response back from the server to see if it includes this header:
1824 Pragma: SuppressEvents
1826 If this is included, the server supports event suppression.
1828 * Added 4 new Actions to list skinny device(s) and line(s)
1834 LDAP Schema File Additions
1835 --------------------------
1836 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
1837 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
1839 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
1840 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
1841 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
1842 * Removed redundant IPaddr (there's already IPAddress)
1843 - Gives more configuration Flags for SIP-Users available (tested)
1844 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
1845 without extensibleObject (which really should be the last resort); gives
1846 also additional possibilities for LDAP-filter
1848 ------------------------------------------------------------------------------
1849 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
1850 ------------------------------------------------------------------------------
1852 Device State Handling
1853 ---------------------
1854 * The event infrastructure in Asterisk got another big update to help support
1855 distributed events. It currently supports distributed device state and
1856 distributed Voicemail MWI (Message Waiting Indication). A new module has
1857 been merged, res_ais, which facilitates communicating events between servers.
1858 It uses the SAForum AIS (Service Availability Forum Application Interface
1859 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
1860 a cluster of Asterisk servers, and to share events between them. For more
1861 information on setting this up, refer to the Distributed Device State section
1862 of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
1866 * Added a new dialplan function, AST_CONFIG(), which allows you to access
1867 variables from an Asterisk configuration file.
1868 * The JACK_HOOK function now has a c() option to supply a custom client name.
1869 * Added two new dialplan functions from libspeex for audio gain control and
1870 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
1871 rx directions of a channel from the dialplan.
1872 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
1873 based on other parameters. The default is still to search based on the
1874 forwarding station ID. However, there are new options that allow you to search
1875 based on the message desk terminal ID, or the message desk number.
1876 * TIMEOUT() has been modified to be accurate down to the millisecond.
1877 * ENUM*() functions now include the following new options:
1878 - 'u' returns the full URI and does not strip off the URI-scheme.
1879 - 's' triggers ISN specific rewriting
1880 - 'i' looks for branches into an Infrastructure ENUM tree
1881 - 'd' for a direct DNS lookup without any flipping of digits.
1882 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
1883 * CHANNEL() now has options for the maximum, minimum, and standard or normal
1884 deviation of jitter, rtt, and loss for a call using chan_sip.
1886 DAHDI channel driver (chan_dahdi) Changes
1887 ----------------------------------------
1888 * Channels can now be configured using named sections in chan_dahdi.conf, just
1889 like other channel drivers, including the use of templates.
1890 * The default for pridialplan has changed from 'national' to 'unknown'.
1894 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
1895 to something that matches the pattern a hint will be created using the contents
1896 and variables evaluated.
1897 * Dialplan matching has been extended to allow an extension to return to the
1898 PBX core to wait for more digits. This is done by using the new dialplan
1899 application called "Incomplete". This will permit a whole new level of
1900 extension control, by giving the administrator more control over early
1901 matches employing one of the short-circuit pattern match operators. Note
1902 that custom applications can trigger this same behavior by returning the
1903 special value AST_PBX_INCOMPLETE.
1907 * Directory now permits both first and last names to be matched at the same
1908 time. In addition, the number of digits to enter of the name can be set in
1909 the arguments to Directory; previously, you could enter only 3, regardless
1910 of how many names are in your company. For large companies, this should be
1912 * Voicemail now permits a mailbox setting to wrap around from first to last
1913 messages, if the "messagewrap" option is set to a true value.
1914 * Voicemail now permits an external script to be run, for password validation.
1915 The script should output "VALID" or "INVALID" on stdout, depending upon the
1916 wish to validate or invalidate the password given. Arguments are:
1917 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
1919 * Dial has a new option: F(context^extension^pri), which permits a callee to
1920 continue in the dialplan, at the specified label, if the caller hangs up.
1921 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
1922 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
1923 * The Jack application now has a c() option to supply a custom client name.
1924 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
1925 like the pre-existing whisper mode, except that the spy can also talk to the
1926 participant on the bridged channel as well.
1927 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
1928 to be spoken instead of the channel name or number. For more information on the
1929 use of this option, issue the command "core show application ChanSpy" from the
1931 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
1932 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
1933 words, if using the 'd' option, it is not possible to enter a number to append to
1934 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
1935 change to whisper mode, and pressing 6 will change to barge mode.
1936 * ExternalIVR now takes several options that affect the way it performs, as
1937 well as having several new commands. Please see the External IVR page on the Asterisk
1938 wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
1939 * Added ability to communicate over a TCP socket instead of forking a child process for the
1940 ExternalIVR application.
1941 * ChanIsAvail has a new option, 'a', which will return all available channels instead
1942 of just the first one if you give the function more then one channel to check.
1943 * PrivacyManager now takes an option where you can specify a context where the
1944 given number will be matched. This way you have more control over who is allowed
1945 and it stops the people who blindly enter 10 digits.
1946 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
1947 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
1948 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
1949 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
1950 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
1951 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
1952 * The Dial() application no longer copies the language used by the caller to the callee's
1953 channel. If you desire for the caller's channel's language to be used for file playback
1954 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
1955 * SendImage() no longer hangs up the channel on error; instead, it sets the
1956 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
1957 'UNSUPPORTED'. This change makes SendImage() more consistent with other
1959 * Park has a new option, 's', which silences the announcement of the parking space number.
1960 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
1961 invalid input and will be assumed to mean that no timeout is desired.
1965 * Added DNS manager support to registrations for peers referencing peer entries.
1966 DNS manager runs in the background which allows DNS lookups to be run asynchronously
1967 as well as periodically updating the IP address. These properties allow for
1968 better performance as well as recovery in the event of an IP change.
1969 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
1970 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
1971 These changes also provide performance improvements for call setup and tear down.
1972 * Added ability to specify registration expiry time on a per registration basis in
1974 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
1976 * Added t38pt_usertpsource option. See sip.conf.sample for details.
1977 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
1978 * 'sip show peers' and 'sip show users' display their entries sorted in
1979 alphabetical order, as opposed to the order they were in, in the config
1981 * Videosupport now supports an additional option, "always", which always sets
1982 up video RTP ports, even on clients that don't support it. This helps with
1983 callfiles and certain transfers to ensure that if two video phones are
1984 connected, they will always share video feeds.
1988 * Existing DNS manager lookups extended to check for SRV records.
1989 * IAX2 encryption support has been improved to support periodic key rotation
1990 within a call for enhanced security. The option "keyrotate" has been
1991 provided to disable this functionality to preserve backwards compatibility
1992 with older versions of IAX2 that do not support key rotation.
1996 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
1997 data tree based on the given <path>.
1998 * New CLI command "data show providers" that will display all the registered
2000 * New CLI command, "config reload <file.conf>" which reloads any module that
2001 references that particular configuration file. Also added "config list"
2002 which shows which configuration files are in use.
2003 * New CLI commands, "pri show version" and "ss7 show version" that will
2004 display which version of libpri and libss7 are being used, respectively.
2005 A new API call was added so trunk will now have to be compiled against
2006 a versions of libpri and libss7 that have them or it will not know that
2007 these libraries exist.
2008 * The commands "core show globals", "core set global" and "core set chanvar" has
2009 been deprecated in favor of the more semanticly correct "dialplan show globals",
2010 "dialplan set chanvar" and "dialplan set global".
2011 * New CLI command "dialplan show chanvar" to list all variables associated
2012 with a given channel.
2016 * Addresses managed by DNS manager now can check to see if there is a DNS
2017 SRV record for a given domain and will use that hostname/port if present.
2019 AMI - The manager (TCP/TLS/HTTP)
2020 --------------------------------
2021 * The Status command now takes an optional list of variables to display
2022 along with channel status.
2023 * The QueueEntry event now also includes the channel's uniqueid
2027 * res_odbc no longer has a limit of 1023 total possible unshared connections,
2028 as some people were running into this limit. This limit has been increased
2033 * The TRANSFER queue log entry now includes the the caller's original
2034 position in the transferred-from queue.
2035 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
2036 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
2037 as well as an explanation about timeout options in general
2038 * Added a new option - C - for forcing the "answered elsewhere" flag on
2039 cancellation of calls in to members of the queue. This is to avoid the
2040 call to a member of a queue having the call listed as a "missed call".
2044 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
2045 adaptive capabilities. What this means in practical terms is that if your
2046 realtime table lacks critical fields, Asterisk will now emit warnings to
2047 that effect. Also, some of the realtime drivers have the ability (if
2048 configured) to automatically add those columns to the table with the
2049 correct type and length.
2053 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
2054 the 'setvar' option to cause a given audio file to be played upon completion
2055 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
2056 Skinny channels only.
2057 * You can now compile Asterisk against the Hoard Memory Allocator, see the
2058 Hoard page on the Asterisk wiki for more information:
2059 https://wiki.asterisk.org/wiki/x/pQBB
2060 * Config file variables may now be appended to, by using the '+=' append
2061 operator. This is most helpful when working with long SQL queries in
2062 func_odbc.conf, as the queries no longer need to be specified on a single
2064 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
2065 which will add a second to the billsec when the ending
2066 time is set, if the number in the microseconds field of the end time is
2067 greater than the number of microseconds in the answer time. This allows
2068 users to count the 'initiated' seconds in their billing records.
2070 ------------------------------------------------------------------------------
2071 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
2072 ------------------------------------------------------------------------------
2074 AMI - The manager (TCP/TLS/HTTP)
2075 --------------------------------
2076 * Manager has undergone a lot of changes, all of them documented
2077 on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
2078 * Manager version has changed to 1.1
2079 * Added a new action 'CoreShowChannels' to list currently defined channels
2080 and some information about them.
2081 * Added a new action 'SIPshowregistry' to list SIP registrations.
2082 * Added TLS support for the manager interface and HTTP server
2083 * Added the URI redirect option for the built-in HTTP server
2084 * The output of CallerID in Manager events is now more consistent.
2085 CallerIDNum is used for number and CallerIDName for name.
2086 * Enable https support for builtin web server.
2087 See configs/http.conf.sample for details.
2088 * Added a new action, GetConfigJSON, which can return the contents of an
2089 Asterisk configuration file in JSON format. This is intended to help
2090 improve the performance of AJAX applications using the manager interface
2092 * SIP and IAX manager events now use "ChannelType" in all cases where we
2093 indicate channel driver. Previously, we used a mixture of "Channel"
2094 and "ChannelDriver" headers.
2095 * Added a "Bridge" action which allows you to bridge any two channels that
2096 are currently active on the system.
2097 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
2098 the voicemail users setup.
2099 * Added 'DBDel' and 'DBDelTree' manager commands.
2100 * cdr_manager now reports events via the "cdr" level, separating it from
2101 the very verbose "call" level.
2102 * Manager users are now stored in memory. If you change the manager account
2103 list (delete or add accounts) you need to reload manager.
2104 * Added Masquerade manager event for when a masquerade happens between
2106 * Added "manager reload" command for the CLI
2107 * Lots of commands that only provided information are now allowed under the
2108 Reporting privilege, instead of only under Call or System.
2109 * The IAX* commands now require either System or Reporting privilege, to
2110 mirror the privileges of the SIP* commands.
2111 * Added ability to retrieve list of categories in a config file.
2112 * Added ability to retrieve the content of a particular category.
2113 * Added ability to empty a context.
2114 * Created new action to create a new file.
2115 * Updated delete action to allow deletion by line number with respect to category.
2116 * Added new action insert to add new variable to category at specified line.
2117 * Updated action newcat to allow new category to be inserted in file above another
2119 * Added new event "JitterBufStats" in the IAX2 channel
2120 * Originate now requires the Originate privilege and, if you want to call out
2121 to a subshell, it requires the System privilege, as well. This was done to
2122 enhance manager security.
2123 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
2124 * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
2125 or manager show command Atxfer from the CLI
2126 * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
2127 details or manager show command IAXregistry from the CLI
2131 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
2132 state in the dialplan, as well as creating custom device states that are
2133 controllable from the dialplan.
2134 * Extend CALLERID() function with "pres" and "ton" parameters to
2135 fetch string representation of calling number presentation indicator
2136 and numeric representation of type of calling number value.
2137 * MailboxExists converted to dialplan function
2138 * A new option to Dial() for telling IP phones not to count the call
2139 as "missed" when dial times out and cancels.
2140 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
2141 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
2142 held for any given channel. Also, locks are automatically freed when a
2144 * Added HINT() dialplan function that allows retrieving hint information.
2145 Hints are mappings between extensions and devices for the sake of
2146 determining the state of an extension. This function can retrieve the list
2147 of devices or the name associated with a hint.
2148 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
2150 * Added SYSINFO() dialplan function which allows retrieval of system information
2151 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
2152 the existence of a dialplan target.
2153 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
2154 upper and lower case, respectively.
2155 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
2156 ID for the call (not the Asterisk call ID or unique ID), provided that the
2157 channel driver supports this. For SIP, you get the SIP call-ID for the
2158 bridged channel which you can store in the CDR with a custom field.
2162 * Added CLI permissions, config file: cli_permissions.conf
2163 default is to allow all commands for every local user/group.
2164 Also this new feature added three new CLI commands:
2165 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
2166 - cli reload permissions
2167 - cli show permissions
2168 * New CLI command "core show hint" (usage: core show hint <exten>)
2169 * New CLI command "core show settings"
2170 * Added 'core show channels count' CLI command.
2171 * Added the ability to set the core debug and verbose values on a per-file basis.
2172 * Added 'queue pause member' and 'queue unpause member' CLI commands
2173 * Ability to set process limits ("ulimit") without restarting Asterisk
2174 * Enhanced "agi debug" to print the channel name as a prefix to the debug
2175 output to make debugging on busy systems much easier.
2176 * New CLI commands "dialplan set extenpatternmatching true/false"
2177 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
2178 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
2179 listed in the startup_commands section of cli.conf will get executed.
2180 * Added a CLI command, "devstate change", which allows you to set custom device
2181 states from the func_devstate module that provides the DEVICE_STATE() function
2182 and handling of the "Custom:" devices.
2183 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
2184 sorted into the different possible callbacks, with the number of entries
2185 currently scheduled for each. Gives you a feel for how busy the sip channel
2187 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
2188 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
2189 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
2193 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
2194 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
2195 for a received call. If it is detected, the channel will jump to the
2196 'fax' extension in the dialplan.
2197 * The default SIP useragent= identifier now includes the Asterisk version
2198 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
2199 If set, and the incoming request carries authentication info,
2200 the username to match in the users list is taken from the Digest header
2201 rather than from the From: field. This feature is considered experimental.
2202 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
2203 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
2204 * The "localmask" setting was removed in version 1.2 and the reminder about it
2205 being removed is now also removed.
2206 * A new option "busylevel" for setting a level of calls where asterisk reports
2207 a device as busy, to separate it from call-limit. This value is also added
2208 to the SIP_PEER dialplan function.
2209 * A new realtime family called "sipregs" is now supported to store SIP registration
2210 data. If this family is defined, "sippeers" will be used for configuration and
2211 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
2212 registration data, as before.
2213 * The SIPPEER function have new options for port address, call and pickup groups
2214 * Added support for T.140 realtime text in SIP/RTP
2215 * The "checkmwi" option has been removed from sip.conf, as it is no longer
2216 required due to the restructuring of how MWI is handled. See the descriptions
2217 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
2218 for more information.
2219 * Added rtpdest option to CHANNEL() dialplan function.
2220 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
2221 * SIP now adds a header to the CANCEL if the call was answered by another phone
2222 in the same dial command, or if the new c option in dial() is used.
2223 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
2224 states it is not needed. For phones, however, that do require it the "registertrying" option
2225 has been added so it can be enabled.
2226 * A new option called "callcounter" (global/peer/user level) enables call counters needed
2227 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
2228 used to enable this functionality).
2229 * New settings for timer T1 and timer B on a global level or per device. This makes it
2230 possible to force timeout faster on non-responsive SIP servers. These settings are
2231 considered advanced, so don't use them unless you have a problem.
2232 * Added a dial string option to be able to set the To: header in an INVITE to any
2234 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
2235 the qualify frequency.
2236 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
2237 were not properly torn down due to network or endpoint failures during an established
2239 * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
2240 and configs/sip.conf.sample for more information on how it is used.
2241 * Added a new configuration option "authfailureevents" that enables manager events when
2242 a peer can't authenticate properly.
2243 * Added DNS manager support to registrations for peers not referencing a peer entry.
2247 * Added the trunkmaxsize configuration option to chan_iax2.
2248 * Added the srvlookup option to iax.conf
2249 * Added support for OSP. The token is set and retrieved through the CHANNEL()
2252 XMPP Google Talk/Jingle changes
2253 -------------------------------
2254 * Added the bindaddr option to gtalk.conf.
2258 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
2259 * Proper codec support in chan_skinny.
2260 * Added settings for IP and Ethernet QoS requests
2264 * Added separate settings for media QoS in mgcp.conf
2266 Console Channel Driver changes
2267 ------------------------------
2268 * Added experimental support for video send & receive to chan_oss.
2269 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
2272 Phone channel changes (chan_phone)
2273 ----------------------------------
2274 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
2276 H.323 channel Changes
2277 ---------------------
2278 * H323 remote hold notification support added (by NOTIFY message
2279 and/or H.450 supplementary service)
2281 Local channel changes
2282 ---------------------
2283 * The device state functionality in the Local channel driver has been updated
2284 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
2285 to just UNKNOWN if the extension exists.
2286 * Added jitterbuffer support for chan_local. This allows you to use the
2287 generic jitterbuffer on incoming calls going to Asterisk applications.
2288 For example, this would allow you to use a jitterbuffer for an incoming
2289 SIP call to Voicemail by putting a Local channel in the middle. This
2290 feature is enabled by using the 'j' option in the Dial string to the Local
2291 channel in conjunction with the existing 'n' option for local channels.
2292 * A 'b' option has been added which causes chan_local to return the actual channel
2293 that is behind it when queried. This is useful for transfer scenarios as the
2294 actual channel will be transferred, not the Local channel.
2296 Agent channel changes
2297 ----------------------
2298 * The ackcall and endcall options are now supplemented with options acceptdtmf
2299 and enddtmf. These allow for the DTMF keypress to be configurable. The options
2300 default to their old hard-coded values ('#' and '*' respectively) so this should
2301 not break any existing agent installations.
2303 DAHDI channel driver (chan_dahdi) Changes
2304 ----------------------------------------
2305 * SS7 support (via libss7 library)
2306 * In India, some carriers transmit CID via dtmf. Some code has been added
2307 that will handle some situations. The cidstart=polarity_IN choice has been added for
2308 those carriers that transmit CID via dtmf after a polarity change.
2309 * CID matching information is now shown when doing 'dialplan show'.
2310 * Added dahdi show version CLI command.
2311 * Added setvar support to chan_dahdi.conf channel entries.
2312 * Added two new options: mwimonitor and mwimonitornotify. These options allow
2313 you to enable MWI monitoring on FXO lines. When the MWI state changes,
2314 the script specified in the mwimonitornotify option is executed. An internal
2315 event indicating the new state of the mailbox is also generated, so that
2316 the normal MWI facilities in Asterisk work as usual.
2317 * Added signalling type 'auto', which attempts to use the same signalling type
2318 for a channel as configured in DAHDI. This is primarily designed for analog
2319 ports, but will also work for digital ports that are configured for FXS or FXO
2320 signalling types. This mode is also the default now, so if your chan_dahdi.conf
2321 does not specify signalling for a channel (which is unlikely as the sample
2322 configuration file has always recommended specifying it for every channel) then
2323 the 'auto' mode will be used for that channel if possible.
2324 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
2325 state for a channel; also ensured that the DNDState Manager event is
2326 emitted no matter how the DND state is set or cleared.
2330 * Added a new channel driver, chan_unistim. See the Asterisk wiki at
2331 https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
2332 for details. This new channel driver allows you to use Nortel i2002,
2333 i2004, and i2050 phones with Asterisk.
2334 * Added a new channel driver, chan_console, which uses portaudio as a cross
2335 platform audio interface. It was written as a channel driver that would
2336 work with Mac CoreAudio, but portaudio supports a number of other audio
2337 interfaces, as well. Note that this channel driver requires v19 or higher
2338 of portaudio; older versions have a different API.
2342 * Added the ability to specify arguments to the Dial application when using
2343 the DUNDi switch in the dialplan.
2344 * Added the ability to set weights for responses dynamically. This can be
2345 done using a global variable or a dialplan function. Using the SHELL()
2346 function would allow you to have an external script set the weight for
2348 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
2349 functions will allow you to initiate a DUNDi query from the dialplan,
2350 find out how many results there are, and access each one.
2351 * Added the ability to specifiy a port for a dundi peer.
2355 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
2356 functions will allow you to initiate an ENUM lookup from the dialplan,
2357 and Asterisk will cache the results. ENUMRESULT can be used to access
2358 the results without doing multiple DNS queries.
2362 * Added the ability to customize which sound files are used for some of the
2363 prompts within the Voicemail application by changing them in voicemail.conf
2364 * Added the ability for the "voicemail show users" CLI command to show users
2365 configured by the dynamic realtime configuration method.
2366 * MWI (Message Waiting Indication) handling has been significantly
2367 restructured internally to Asterisk. It is now totally event based
2368 instead of polling based. The voicemail application will notify other
2369 modules that have subscribed to MWI events when something in the mailbox
2371 This also means that if any other entity outside of Asterisk is changing
2372 the contents of mailboxes, then the voicemail application still needs to
2373 poll for changes. Examples of situations that would require this option
2374 are web interfaces to voicemail or an email client in the case of using
2375 IMAP storage. So, two new options have been added to voicemail.conf
2376 to account for this: "pollmailboxes" and "pollfreq". See the sample
2377 configuration file for details.
2378 * Added "tw" language support
2379 * Added support for storage of greetings using an IMAP server
2380 * Added ability to customize forward, reverse, stop, and pause keys for message playback
2381 * SMDI is now enabled in voicemail using the smdienable option.
2382 * A "lockmode" option has been added to asterisk.conf to configure the file
2383 locking method used for voicemail, and potentially other things in the
2384 future. The default is the old behavior, lockfile. However, there is a
2385 new method, "flock", that uses a different method for situations where the
2386 lockfile will not work, such as on SMB/CIFS mounts.
2387 * Added the ability to backup deleted messages, to ease recovery in the case
2388 that a user accidentally deletes a message, and discovers that they need it.
2389 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
2390 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
2391 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
2392 voicemail boxes. The SMDI interface can also poll for MWI changes when some
2393 outside entity is modifying the state of the mailbox (such as IMAP storage or
2394 a web interface of some kind).
2395 * Added the support for marking messages as "urgent." There are two methods to accomplish
2396 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
2397 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
2398 the message as urgent after he has recorded a voicemail by following the voice instructions.
2399 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
2404 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
2405 used across multiple queues.
2406 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
2407 setqueueentryvar options for each queue, see queues.conf.sample for details.
2408 * Added keepstats option to queues.conf which will keep queue
2409 statistics during a reload.
2410 * setinterfacevar option in queues.conf also now sets a variable
2411 called MEMBERNAME which contains the member's name.
2412 * Added 'Strategy' field to manager event QueueParams which represents
2413 the queue strategy in use.
2414 * Added option to run macro when a queue member is connected to a caller,
2415 see queues.conf.sample for details.
2416 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
2417 does not count paused queue members as unavailable.
2418 * Added min-announce-frequency option to queues.conf which allows you to control the
2419 minimum amount of time between queue announcements for use when the caller's queue
2420 position changes frequently.
2421 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
2423 * Added ability for non-realtime queues to have realtime members
2424 * Added the "linear" strategy to queues.
2425 * Added the "wrandom" strategy to queues.
2426 * Added new channel variable QUEUE_MIN_PENALTY
2427 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
2428 rules in queuerules.conf. See configs/queuerules.conf.sample for details
2429 * Added a new parameter for member definition, called state_interface. This may be
2430 used so that a member may be called via one interface but have a different interface's
2431 device state reported.
2432 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
2433 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
2434 "manager show command QueueReset."
2435 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
2436 specified by the periodic-announce option, then one will be chosen randomly when it is time
2437 to play a periodic announcment
2438 * New configuration options: announce-position now takes two more values in addition to "yes" and
2439 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
2440 announce-position-limit. By setting announce-position to "limit" callers will only have their
2441 position announced if their position is less than what is specified by announce-position-limit.
2442 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
2443 will be told that their are more than announce-position-limit callers waiting.
2444 * Two new queue log events have been added. An ADDMEMBER event will be logged
2445 when a realtime queue member is added and a REMOVEMEMBER event will be logged
2446 when a realtime queue member is removed. Since there is no calling channel associated
2447 with these events, the string "REALTIME" is placed where the channel's unique id
2448 is typically placed.
2449 * The configuration method for the "joinempty" and "leavewhenempty" options has
2450 changed to a comma-separated list of methods of determining member availability
2451 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
2452 values are still accepted for backwards-compatibility, though.
2453 * The average talktime is now calculated on queues. This information is reported via the
2454 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
2455 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
2460 * The 'o' option to provide an optimization has been removed and its functionality
2461 has been enabled by default.
2462 * When a conference is created, the UNIQUEID of the channel that caused it to be
2463 created is stored. Then, every channel that joins the conference will have the
2464 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
2465 callers that come and go from long standing conferences.
2466 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
2467 except it does operations on a channel by name, instead of number in a conference.
2468 This is a very useful feature in combination with the 'X' option to ChanSpy.
2469 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
2471 * Added new RealTime functionality to provide support for scheduled conferencing.
2472 This includes optional messages to the caller if they attempt to join before
2473 the schedule start time, or to allow the caller to join the conference early.
2474 Also included is optional support for limiting the number of callers per
2475 RealTime conference.
2476 * Added the S() and L() options to the MeetMe application. These are pretty
2477 much identical to the S() and L() options to Dial(). They let you set
2478 timeouts for the conference, as well as have warning sounds played to
2479 let the caller know how much time is left, and when it is running out.
2480 * Added the ability to do "meetme concise" with the "meetme" CLI command.
2481 This extends the concise capabilities of this CLI command to include
2482 listing all conferences, instead of an addition to the other sub commands
2483 for the "meetme" command.
2484 * Added the ability to specify the music on hold class used to play into the
2485 conference when there is only one member and the M option is used.
2486 * Added MEETME_INFO dialplan function which provides a way to query
2487 various properties of a Meetme conference.
2488 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
2489 and *84: record in-conf
2491 Other Dialplan Application Changes
2492 ----------------------------------
2493 * Argument support for Gosub application
2494 * From the to-do lists: straighten out the app timeout args:
2495 Wait() app now really does 0.3 seconds- was truncating arg to an int.
2496 WaitExten() same as Wait().
2497 Congestion() - Now takes floating pt. argument.
2498 Busy() - now takes floating pt. argument.
2499 Read() - timeout now can be floating pt.
2500 WaitForRing() now takes floating pt timeout arg.
2501 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
2502 * Added 's' option to Page application.
2503 * Added an optional timeout argument to the Page application.
2504 * Added 'E', 'V', and 'P' commands to ExternalIVR.
2505 * Added 'o' and 'X' options to Chanspy.
2506 * Added a new dialplan application, Bridge, which allows you to bridge the
2507 calling channel to any other active channel on the system.
2508 * Added the ability to specify a music on hold class to play instead of ringing
2509 for the SLATrunk application.
2510 * The Read application no longer exits the dialplan on error. Instead, it sets
2511 READSTATUS to ERROR, which you can catch and handle separately.
2512 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
2513 of asking for verification of each name, one at a time.
2514 * Privacy() no longer uses privacy.conf, as all options are specifyable as
2515 direct options to the app.
2516 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
2518 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
2519 * The ChannelRedirect application no longer exits the dialplan if the given channel
2520 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
2521 or NOCHANNEL if the given channel was not found.
2522 * The silencethreshold setting that was previously configurable in multiple
2523 applications is now settable globally via dsp.conf.
2525 Music On Hold Changes
2526 ---------------------
2527 * A new option, "digit", has been added for music on hold classes in
2528 musiconhold.conf. If this is set for a music on hold class, a caller
2529 listening to music on hold can press this digit to switch to listening
2530 to this music on hold class.
2531 * Support for realtime music on hold has been added.
2532 * In conjunction with the realtime music on hold, a general section has
2533 been added to musiconhold.conf, its sole variable is cachertclasses. If this
2534 is set, then music on hold classes found in realtime will be cached in memory.
2538 * AEL upgraded to use the Gosub with Arguments instead
2539 of Macro application, to hopefully reduce the problems
2540 seen with the artificially low stack ceiling that
2541 Macro bumps into. Macros can only call other Macros
2542 to a depth of 7. Tests run using gosub, show depths
2543 limited only by virtual memory. A small test demonstrated
2544 recursive call depths of 100,000 without problems.
2545 -- in addition to this, all apps that allowed a macro
2546 to be called, as in Dial, queues, etc, are now allowing
2547 a gosub call in similar fashion.
2548 * AEL now generates LOCAL(argname) declarations when it
2549 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
2550 etc. That makes the arguments local in scope. The user
2551 can define their own local variables in macros, now,
2552 by saying "local myvar=someval;" or using Set() in this
2553 fashion: Set(LOCAL(myvar)=someval); ("local" is now
2555 * utils/conf2ael introduced. Will convert an extensions.conf
2556 file into extensions.ael. Very crude and unfinished, but
2557 will be improved as time goes by. Should be useful for a
2558 first pass at conversion.
2559 * aelparse will now read extensions.conf to see if a referenced
2560 macro or context is there before issueing a warning.
2561 * AEL parser sets a local channel variable ~~EXTEN~~, to
2562 preserve the value of ${EXTEN} thru switch statements.
2563 * New operator in $[...] expressions: the ~~ operator serves
2564 as a concatenation operator. AT THE MOMENT, it is really only
2565 necessary and useful in AEL, especially in if() expressions.
2566 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
2567 any enclosing double-quotes, and evaluate to the value of a
2568 concatenated with the value of b. For example if a is set to
2569 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
2570 evaluate to xyzabc .
2573 Call Features (res_features) Changes
2574 ------------------------------------
2575 * Added the parkedcalltransfers option to features.conf
2576 * Added parkedcallparking option to control one touch parking w/ parking
2578 * Added parkedcallhangup option to control disconnect feature w/ parking
2580 * Added parkedcallrecording option to control one-touch record w/ parking
2582 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
2583 parkedcalltransfers option support for multiple parking lots.
2584 * Added BRIDGE_FEATURES variable to set available features for a channel
2585 * The built-in method for doing attended transfers has been updated to
2586 include some new options that allow you to have the transferee sent
2587 back to the person that did the transfer if the transfer is not successful.
2588 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
2589 in features.conf.sample.
2590 * Added support for configuring named groups of custom call features in
2591 features.conf. This means that features can be written a single time, and
2592 then mapped into groups of features for different key mappings or easier
2594 * Updated the ParkedCall application to allow you to not specify a parking
2595 extension. If you don't specify a parking space to pick up, it will grab
2596 the first one available.
2597 * Added cli command 'features reload' to reload call features from features.conf
2598 * Moved into core asterisk binary.
2599 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
2600 * Added the ability for custom parking lots to be configured with their own
2601 parking extension with the parkext option.
2603 Language Support Changes
2604 ------------------------
2605 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
2606 * Added support for the Hungarian language for saying numbers, dates, and times.
2610 * Added SPEECH commands for speech recognition. A complete listing can be found
2612 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
2613 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
2614 does not behave as expected; the native command needs to be used, instead.
2615 * Added the ability to perform SRV lookups on fast AGI calls. To use this
2616 feature, simply use hagi: instead of agi: as the protocol portion
2617 of the URI parameter to the AGI function call in your dial plan. Also note
2618 that specifying a port number in the AGI URI will disable SRV lookups,
2619 even if you use the hagi: protocol.
2620 * No longer support MSG_OOB flag on HANGUP.
2624 * Added rotatestrategy option to logger.conf, along with two new options:
2625 "timestamp" which will use the time to name the logger files instead of
2626 sequence number; and "rotate", which rotates the names of the log files,
2627 similar to the way syslog rotates files.
2628 * Added exec_after_rotate option to logger.conf, which allows a system
2629 command to be run after rotation. This is primarily useful with
2630 rotatestrategy=rotate, to allow a limit on the number of log files kept
2631 and to ensure that the oldest log file gets deleted.
2632 * Added realtime support for the queue log
2636 * The cdr_manager module has a [mappings] feature, like cdr_custom,
2637 to add fields to the manager event from the CDR variables.
2638 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
2639 backend database CDR table. Specifically, additional, non-standard
2640 columns are supported, merely by setting the corresponding CDR variable in
2641 your dialplan. In addition, you may alias any column to another name (for
2642 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
2643 simply "alias src => ANI" in the configuration file). Records may be
2644 posted to more than one backend, simply by specifying multiple categories
2645 in the configuration file. And finally, you may filter which CDRs get
2646 posted to each backend, by specifying a filter (which the record must
2647 match) for the particular category. Filters are additive (meaning all
2648 rules must match to post that CDR).
2649 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
2650 module. Specifically, you may add additional columns into the table and
2651 they will be set, if you set the corresponding CDR variable name. Also,
2652 if you omit columns in your database table, they will be silently skipped
2653 (but a record will still be inserted, based on what columns remain). Note
2654 that the other two features from cdr_adaptive_odbc (alias and filter) are
2655 not currently supported.
2656 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
2657 has been disabled using the NoCDR application.
2659 Miscellaneous New Modules
2660 -------------------------
2661 * Added a new CDR module, cdr_sqlite3_custom.
2662 * Added a new realtime configuration module, res_config_sqlite
2663 * Added a new codec translation module, codec_resample, which re-samples
2664 signed linear audio between 8 kHz and 16 kHz to help support wideband
2666 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
2667 based on configuration templates that use Asterisk dialplan function and
2668 variable substitution. It should be possible to create phone profiles and
2669 templates that work for the majority of phones provisioned over http. It
2670 is currently only intended to provision a single user account per phone.
2671 An example profile and set of templates for Polycom phones is provided.
2672 NOTE: Polycom firmware is not included, but should be placed in
2673 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
2674 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
2675 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
2676 provided; there is a JACK() application, and a JACK_HOOK() function. Both
2677 interfaces create an input and output JACK port. The application makes
2678 these ports the endpoint of the call. The audio coming from the channel
2679 goes out the output port and whatever comes back in on the input port is
2680 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
2681 audiohook on the channel. This lets you run the audio coming from a
2682 channel through JACK, and whatever comes back in is what gets forwarded
2683 on as the channel's audio. This is very useful for building custom
2684 vocoders or doing recording or analysis of the channel's audio in another
2686 * Added a new module, res_config_curl, which permits using a HTTP POST url
2687 to retrieve, create, update, and delete realtime information from a remote
2688 web server. Note that this module requires func_curl.so to be loaded for
2689 backend functionality.
2690 * Added a new module, res_config_ldap, which permits the use of an LDAP
2691 server for realtime data access.
2692 * Added support for writing and running your dialplan in lua using the pbx_lua
2693 module. See configs/extensions.lua.sample for examples of how to do this.
2697 * Ability to use libcap to set high ToS bits when non-root
2698 on Linux. If configure is unable to find libcap then you
2699 can use --with-cap to specify the path.
2700 * Added maxfiles option to options section of asterisk.conf which allows you to specify
2701 what Asterisk should set as the maximum number of open files when it loads.
2702 * Added the jittertargetextra configuration option.
2703 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
2704 configuration files for the IP channel drivers. The new option is "cos".
2705 This information is also documented on the Asterisk wiki at
2706 https://wiki.asterisk.org/wiki/x/EYBG
2707 * When originating a call using AMI or pbx_spool that fails the reason for failure
2708 will now be available in the failed extension using the REASON dialplan variable.
2709 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
2710 It allows you to configure a prefix for auto-monitor recordings.
2711 * A new extension pattern matching algorithm, based on a trie, is introduced
2712 here, that could noticeably speed up mid-sized to large dialplans.
2713 It is NOT used by default, as duplicating the behaviour of the old pattern
2714 matcher is still under development. A config file option, in extensions.conf,
2715 in the [general] section, called "extenpatternmatchingnew", is by default
2716 set to false; setting that to true will force the use of the new algorithm.
2717 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
2718 be used to switch the algorithms at run time.
2719 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
2720 specifying which socket to use to connect to the running Asterisk daemon
2722 * Performance enhancements to the sched facility, which is used in
2723 the channel drivers, etc. Added hashtabs and doubly-linked lists
2724 to speed up deletion; start at the beginning or end of list to
2726 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
2727 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
2728 Added regression tests to the tests/ dir, also.
2729 * Added a refcount trace feature to astobj2 for those trying to balance
2730 object creation, deletion; work, play; space and time. See the
2731 notes in astobj2.h. Also, see utils/refcounter as well, as a
2732 quick way to find unbalanced refcounts in what could be a sea
2733 of objects that were balanced.
2734 * Added logging to 'make update' command. See update.log
2735 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
2736 do not come from the remote party.
2737 * Added the 'n' option to the SpeechBackground application to tell it to not
2738 answer the channel if it has not already been answered.
2739 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
2740 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
2742 * iLBC source code no longer included (see UPGRADE.txt for details)
2743 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
2744 deadlock is detected, a backtrace of the stack which led to the lock calls
2745 will be output to the CLI.
2746 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
2747 the "core show locks" CLI command will give lock information output as well
2748 as a backtrace of the stack which led to the lock calls.
2749 * users.conf now sports an optional alternateexts property, which permits
2750 allocation of additional extensions which will reach the specified user.
2751 * A new option for the configure script, --enable-internal-poll, has been added
2752 for use with systems which may have a buggy implementation of the poll system
2753 call. If you notice odd behavior such as the CLI being unresponsive on remote
2754 consoles, you may want to try using this option. This option is enabled by default
2755 on Darwin systems since it is known that the Darwin poll() implementation has
2759 --------------------
2760 * In addition to timing from DAHDI, there is a new timing module called
2761 res_timing_timerfd. In order to use this, you must be running Linux with
2762 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
2763 script will be able to tell if you have the requirements. From menuselect, select
2764 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob