| blob | ef3f769c978f0050b9a852500dd0e7cf1037ee4c |
1 ------------------------------------------------------------------------------
2 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
3 ------------------------------------------------------------------------------
5 Device State Handling
6 ---------------------
7 * The event infrastructure in Asterisk got another big update to help support
8 distributed events. It currently supports distributed device state and
9 distributed Voicemail MWI (Message Waiting Indication). A new module has
10 been merged, res_ais, which facilitates communicating events between servers.
11 It uses the SAForum AIS (Service Availability Forum Application Interface
12 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
13 a cluster of Asterisk servers, and to share events between them. For more
14 information on setting this up, see doc/distributed_devstate.txt.
16 Dialplan Functions
17 ------------------
18 * Added a new dialplan function, AST_CONFIG(), which allows you to access
19 variables from an Asterisk configuration file.
20 * The JACK_HOOK function now has a c() option to supply a custom client name.
21 * Added two new dialplan functions from libspeex for audio gain control and
22 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
23 rx directions of a channel from the dialplan.
24 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
25 based on other parameters. The default is still to search based on the
26 forwarding station ID. However, there are new options that allow you to search
27 based on the message desk terminal ID, or the message desk number.
28 * TIMEOUT() has been modified to be accurate down to the millisecond.
29 * ENUM*() functions now include the following new options:
30 - 'u' returns the full URI and does not strip off the URI-scheme.
31 - 's' triggers ISN specific rewriting
32 - 'i' looks for branches into an Infrastructure ENUM tree
33 - 'd' for a direct DNS lookup without any flipping of digits.
34 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
35 * CHANNEL() now has options for the maximum, minimum, and standard or normal
36 deviation of jitter, rtt, and loss for a call using chan_sip.
38 DAHDI channel driver (chan_dahdi) Changes
39 ----------------------------------------
40 * Channels can now be configured using named sections in chan_dahdi.conf, just
41 like other channel drivers, including the use of templates.
42 * The default for pridialplan has changed from 'national' to 'unknown'.
44 PBX Changes
45 -----------
46 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
47 to something that matches the pattern a hint will be created using the contents
48 and variables evaluated.
49 * Dialplan matching has been extended to allow an extension to return to the
50 PBX core to wait for more digits. This is done by using the new dialplan
51 application called "Incomplete". This will permit a whole new level of
52 extension control, by giving the administrator more control over early
53 matches employing one of the short-circuit pattern match operators. Note
54 that custom applications can trigger this same behavior by returning the
55 special value AST_PBX_INCOMPLETE.
57 Application Changes
58 -------------------
59 * Directory now permits both first and last names to be matched at the same
60 time. In addition, the number of digits to enter of the name can be set in
61 the arguments to Directory; previously, you could enter only 3, regardless
62 of how many names are in your company. For large companies, this should be
63 quite helpful.
64 * Voicemail now permits a mailbox setting to wrap around from first to last
65 messages, if the "messagewrap" option is set to a true value.
66 * Voicemail now permits an external script to be run, for password validation.
67 The script should output "VALID" or "INVALID" on stdout, depending upon the
68 wish to validate or invalidate the password given. Arguments are:
69 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
70 more details
71 * Dial has a new option: F(context^extension^pri), which permits a callee to
72 continue in the dialplan, at the specified label, if the caller hangs up.
73 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
74 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
75 * The Jack application now has a c() option to supply a custom client name.
76 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
77 like the pre-existing whisper mode, except that the spy can also talk to the
78 participant on the bridged channel as well.
79 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
80 to be spoken instead of the channel name or number. For more information on the
81 use of this option, issue the command "core show application ChanSpy" from the
82 Asterisk CLI.
83 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
84 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
85 words, if using the 'd' option, it is not possible to enter a number to append to
86 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
87 change to whisper mode, and pressing 6 will change to barge mode.
88 * ExternalIVR now takes several options that affect the way it performs, as
89 well as having several new commands. Please see doc/externalivr.txt for the
90 complete documentation.
91 * ChanIsAvail has a new option, 'a', which will return all available channels instead
92 of just the first one if you give the function more then one channel to check.
93 * PrivacyManager now takes an option where you can specify a context where the
94 given number will be matched. This way you have more control over who is allowed
95 and it stops the people who blindly enter 10 digits.
96 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
97 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
98 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
99 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
100 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
101 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
102 * The Dial() application no longer copies the language used by the caller to the callee's
103 channel. If you desire for the caller's channel's language to be used for file playback
104 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
105 * SendImage() no longer hangs up the channel on error; instead, it sets the
106 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
107 'UNSUPPORTED'. This change makes SendImage() more consistent with other
108 applications.
109 * Park has a new option, 's', which silences the announcement of the parking space number.
111 SIP Changes
112 -----------
113 * Added DNS manager support to registrations for peers referencing peer entries.
114 DNS manager runs in the background which allows DNS lookups to be run asynchronously
115 as well as periodically updating the IP address. These properties allow for
116 better performance as well as recovery in the event of an IP change.
117 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
118 load/reload of large numbers of peers/users by ~40x (for large lists of peers.
119 Initially, we saw 4x improvement in call setup/destruction, but at the time
120 of merging, this gain has disappeared; further research will be done to try
121 and restore this performance improvement. Astobj2 refcounting is now used
122 for users, peers, and dialogs. Users are encouraged to assist in regression
123 testing and problem reporting!
124 * Added ability to specify registration expiry time on a per registration basis in
125 the register line.
126 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
127 lost packets.
128 * Added t38pt_usertpsource option. See sip.conf.sample for details.
129 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
130 * 'sip show peers' and 'sip show users' display their entries sorted in
131 alphabetical order, as opposed to the order they were in, in the config
132 file or database.
133 * Videosupport now supports an additional option, "always", which always sets
134 up video RTP ports, even on clients that don't support it. This helps with
135 callfiles and certain transfers to ensure that if two video phones are
136 connected, they will always share video feeds.
138 IAX Changes
139 -----------
140 * Existing DNS manager lookups extended to check for SRV records.
141 * IAX2 encryption support has been improved to support periodic key rotation
142 within a call for enhanced security. The option "keyrotate" has been
143 provided to disable this functionality to preserve backwards compatibility
144 with older versions of IAX2 that do not support key rotation.
146 CLI Changes
147 -----------
148 * New CLI command, "config reload <file.conf>" which reloads any module that
149 references that particular configuration file. Also added "config list"
150 which shows which configuration files are in use.
151 * New CLI commands, "pri show version" and "ss7 show version" that will
152 display which version of libpri and libss7 are being used, respectively.
153 A new API call was added so trunk will now have to be compiled against
154 a versions of libpri and libss7 that have them or it will not know that
155 these libraries exist.
156 * The commands "core show globals", "core set global" and "core set chanvar" has
157 been deprecated in favor of the more semanticly correct "dialplan show globals",
158 "dialplan set chanvar" and "dialplan set global".
159 * New CLI command "dialplan show chanvar" to list all variables associated
160 with a given channel.
162 DNS manager changes
163 -------------------
164 * Addresses managed by DNS manager now can check to see if there is a DNS
165 SRV record for a given domain and will use that hostname/port if present.
167 AMI - The manager (TCP/TLS/HTTP)
168 --------------------------------
169 * The Status command now takes an optional list of variables to display
170 along with channel status.
172 ODBC Changes
173 ------------
174 * res_odbc no longer has a limit of 1023 total possible unshared connections,
175 as some people were running into this limit. This limit has been increased
176 to 4.2 billion.
178 Queue changes
179 -------------
180 * The TRANSFER queue log entry now includes the the caller's original
181 position in the transferred-from queue.
182 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
183 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
184 as well as an explanation about timeout options in general
186 Realtime changes
187 ----------------
188 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
189 adaptive capabilities. What this means in practical terms is that if your
190 realtime table lacks critical fields, Asterisk will now emit warnings to
191 that effect. Also, some of the realtime drivers have the ability (if
192 configured) to automatically add those columns to the table with the
193 correct type and length.
195 Miscellaneous
196 -------------
197 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
198 the 'setvar' option to cause a given audio file to be played upon completion
199 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
200 Skinny channels only.
201 * You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt
202 for more information.
203 * Config file variables may now be appended to, by using the '+=' append
204 operator. This is most helpful when working with long SQL queries in
205 func_odbc.conf, as the queries no longer need to be specified on a single
206 line.
208 ------------------------------------------------------------------------------
209 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
210 ------------------------------------------------------------------------------
212 AMI - The manager (TCP/TLS/HTTP)
213 --------------------------------
214 * Manager has undergone a lot of changes, all of them documented
215 in doc/manager_1_1.txt
216 * Manager version has changed to 1.1
217 * Added a new action 'CoreShowChannels' to list currently defined channels
218 and some information about them.
219 * Added a new action 'SIPshowregistry' to list SIP registrations.
220 * Added TLS support for the manager interface and HTTP server
221 * Added the URI redirect option for the built-in HTTP server
222 * The output of CallerID in Manager events is now more consistent.
223 CallerIDNum is used for number and CallerIDName for name.
224 * Enable https support for builtin web server.
225 See configs/http.conf.sample for details.
226 * Added a new action, GetConfigJSON, which can return the contents of an
227 Asterisk configuration file in JSON format. This is intended to help
228 improve the performance of AJAX applications using the manager interface
229 over HTTP.
230 * SIP and IAX manager events now use "ChannelType" in all cases where we
231 indicate channel driver. Previously, we used a mixture of "Channel"
232 and "ChannelDriver" headers.
233 * Added a "Bridge" action which allows you to bridge any two channels that
234 are currently active on the system.
235 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
236 the voicemail users setup.
237 * Added 'DBDel' and 'DBDelTree' manager commands.
238 * cdr_manager now reports events via the "cdr" level, separating it from
239 the very verbose "call" level.
240 * Manager users are now stored in memory. If you change the manager account
241 list (delete or add accounts) you need to reload manager.
242 * Added Masquerade manager event for when a masquerade happens between
243 two channels.
244 * Added "manager reload" command for the CLI
245 * Lots of commands that only provided information are now allowed under the
246 Reporting privilege, instead of only under Call or System.
247 * The IAX* commands now require either System or Reporting privilege, to
248 mirror the privileges of the SIP* commands.
249 * Added ability to retrieve list of categories in a config file.
250 * Added ability to retrieve the content of a particular category.
251 * Added ability to empty a context.
252 * Created new action to create a new file.
253 * Updated delete action to allow deletion by line number with respect to category.
254 * Added new action insert to add new variable to category at specified line.
255 * Updated action newcat to allow new category to be inserted in file above another
256 existing category.
257 * Added new event "JitterBufStats" in the IAX2 channel
258 * Originate now requires the Originate privilege and, if you want to call out
259 to a subshell, it requires the System privilege, as well. This was done to
260 enhance manager security.
261 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
262 * New command: Atxfer. See doc/manager_1_1.txt for more details or
263 manager show command Atxfer from the CLI
265 Dialplan functions
266 ------------------
267 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
268 state in the dialplan, as well as creating custom device states that are
269 controllable from the dialplan.
270 * Extend CALLERID() function with "pres" and "ton" parameters to
271 fetch string representation of calling number presentation indicator
272 and numeric representation of type of calling number value.
273 * MailboxExists converted to dialplan function
274 * A new option to Dial() for telling IP phones not to count the call
275 as "missed" when dial times out and cancels.
276 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
277 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
278 held for any given channel. Also, locks are automatically freed when a
279 channel is hung up.
280 * Added HINT() dialplan function that allows retrieving hint information.
281 Hints are mappings between extensions and devices for the sake of
282 determining the state of an extension. This function can retrieve the list
283 of devices or the name associated with a hint.
284 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
285 of any extension.
286 * Added SYSINFO() dialplan function which allows retrieval of system information
287 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
288 the existence of a dialplan target.
289 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
290 upper and lower case, respectively.
291 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
292 ID for the call (not the Asterisk call ID or unique ID), provided that the
293 channel driver supports this. For SIP, you get the SIP call-ID for the
294 bridged channel which you can store in the CDR with a custom field.
296 CLI Changes
297 -----------
298 * New CLI command "core show hint" (usage: core show hint <exten>)
299 * New CLI command "core show settings"
300 * Added 'core show channels count' CLI command.
301 * Added the ability to set the core debug and verbose values on a per-file basis.
302 * Added 'queue pause member' and 'queue unpause member' CLI commands
303 * Ability to set process limits ("ulimit") without restarting Asterisk
304 * Enhanced "agi debug" to print the channel name as a prefix to the debug
305 output to make debugging on busy systems much easier.
306 * New CLI commands "dialplan set extenpatternmatching true/false"
307 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
308 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
309 listed in the startup_commands section of cli.conf will get executed.
310 * Added a CLI command, "devstate change", which allows you to set custom device
311 states from the func_devstate module that provides the DEVICE_STATE() function
312 and handling of the "Custom:" devices.
313 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
314 sorted into the different possible callbacks, with the number of entries
315 currently scheduled for each. Gives you a feel for how busy the sip channel
316 driver is.
318 SIP changes
319 -----------
320 * Improved NAT and STUN support.
321 chan_sip now can use port numbers in bindaddr, externip and externhost
322 options, as well as contact a STUN server to detect its external address
323 for the SIP socket. See sip.conf.sample, 'NAT' section.
324 * The default SIP useragent= identifier now includes the Asterisk version
325 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
326 If set, and the incoming request carries authentication info,
327 the username to match in the users list is taken from the Digest header
328 rather than from the From: field. This feature is considered experimental.
329 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
330 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
331 * The "localmask" setting was removed in version 1.2 and the reminder about it
332 being removed is now also removed.
333 * A new option "busylevel" for setting a level of calls where asterisk reports
334 a device as busy, to separate it from call-limit. This value is also added
335 to the SIP_PEER dialplan function.
336 * A new realtime family called "sipregs" is now supported to store SIP registration
337 data. If this family is defined, "sippeers" will be used for configuration and
338 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
339 registration data, as before.
340 * The SIPPEER function have new options for port address, call and pickup groups
341 * Added support for T.140 realtime text in SIP/RTP
342 * The "checkmwi" option has been removed from sip.conf, as it is no longer
343 required due to the restructuring of how MWI is handled. See the descriptions
344 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
345 for more information.
346 * Added rtpdest option to CHANNEL() dialplan function.
347 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
348 * SIP now adds a header to the CANCEL if the call was answered by another phone
349 in the same dial command, or if the new c option in dial() is used.
350 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
351 states it is not needed. For phones, however, that do require it the "registertrying" option
352 has been added so it can be enabled.
353 * A new option called "callcounter" (global/peer/user level) enables call counters needed
354 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
355 used to enable this functionality).
356 * New settings for timer T1 and timer B on a global level or per device. This makes it
357 possible to force timeout faster on non-responsive SIP servers. These settings are
358 considered advanced, so don't use them unless you have a problem.
359 * Added a dial string option to be able to set the To: header in an INVITE to any
360 SIP uri.
361 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
362 the qualify frequency.
363 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
364 were not properly torn down due to network or endpoint failures during an established
365 SIP session.
366 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
367 configs/sip.conf.sample for more information on how it is used.
368 * Added a new configuration option "authfailureevents" that enables manager events when
369 a peer can't authenticate properly.
370 * Added DNS manager support to registrations for peers not referencing a peer entry.
372 IAX2 changes
373 ------------
374 * Added the trunkmaxsize configuration option to chan_iax2.
375 * Added the srvlookup option to iax.conf
376 * Added support for OSP. The token is set and retrieved through the CHANNEL()
377 dialplan function.
379 XMPP Google Talk/Jingle changes
380 -------------------------------
381 * Added the bindaddr option to gtalk.conf.
383 Skinny changes
384 -------------
385 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
386 * Proper codec support in chan_skinny.
387 * Added settings for IP and Ethernet QoS requests
389 MGCP changes
390 ------------
391 * Added separate settings for media QoS in mgcp.conf
393 Console Channel Driver changes
394 ------------------------------
395 * Added experimental support for video send & receive to chan_oss.
396 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
397 a video source.
399 Phone channel changes (chan_phone)
400 ----------------------------------
401 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
403 H.323 channel Changes
404 ---------------------
405 * H323 remote hold notification support added (by NOTIFY message
406 and/or H.450 supplementary service)
408 Local channel changes
409 ---------------------
410 * The device state functionality in the Local channel driver has been updated
411 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
412 to just UNKNOWN if the extension exists.
413 * Added jitterbuffer support for chan_local. This allows you to use the
414 generic jitterbuffer on incoming calls going to Asterisk applications.
415 For example, this would allow you to use a jitterbuffer for an incoming
416 SIP call to Voicemail by putting a Local channel in the middle. This
417 feature is enabled by using the 'j' option in the Dial string to the Local
418 channel in conjunction with the existing 'n' option for local channels.
419 * A 'b' option has been added which causes chan_local to return the actual channel
420 that is behind it when queried. This is useful for transfer scenarios as the
421 actual channel will be transferred, not the Local channel.
423 Agent channel changes
424 ----------------------
425 * The ackcall and endcall options are now supplemented with options acceptdtmf
426 and enddtmf. These allow for the DTMF keypress to be configurable. The options
427 default to their old hard-coded values ('#' and '*' respectively) so this should
428 not break any existing agent installations.
430 DAHDI channel driver (chan_dahdi) Changes
431 ----------------------------------------
432 * SS7 support (via libss7 library)
433 * In India, some carriers transmit CID via dtmf. Some code has been added
434 that will handle some situations. The cidstart=polarity_IN choice has been added for
435 those carriers that transmit CID via dtmf after a polarity change.
436 * CID matching information is now shown when doing 'dialplan show'.
437 * Added dahdi show version CLI command.
438 * Added setvar support to chan_dahdi.conf channel entries.
439 * Added two new options: mwimonitor and mwimonitornotify. These options allow
440 you to enable MWI monitoring on FXO lines. When the MWI state changes,
441 the script specified in the mwimonitornotify option is executed. An internal
442 event indicating the new state of the mailbox is also generated, so that
443 the normal MWI facilities in Asterisk work as usual.
444 * Added signalling type 'auto', which attempts to use the same signalling type
445 for a channel as configured in DAHDI. This is primarily designed for analog
446 ports, but will also work for digital ports that are configured for FXS or FXO
447 signalling types. This mode is also the default now, so if your chan_dahdi.conf
448 does not specify signalling for a channel (which is unlikely as the sample
449 configuration file has always recommended specifying it for every channel) then
450 the 'auto' mode will be used for that channel if possible.
451 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
452 state for a channel; also ensured that the DNDState Manager event is
453 emitted no matter how the DND state is set or cleared.
455 New Channel Drivers
456 -------------------
457 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
458 configs/unistim.conf.sample for details. This new channel driver allows
459 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
460 * Added a new channel driver, chan_console, which uses portaudio as a cross
461 platform audio interface. It was written as a channel driver that would
462 work with Mac CoreAudio, but portaudio supports a number of other audio
463 interfaces, as well. Note that this channel driver requires v19 or higher
464 of portaudio; older versions have a different API.
466 DUNDi changes
467 -------------
468 * Added the ability to specify arguments to the Dial application when using
469 the DUNDi switch in the dialplan.
470 * Added the ability to set weights for responses dynamically. This can be
471 done using a global variable or a dialplan function. Using the SHELL()
472 function would allow you to have an external script set the weight for
473 each response.
474 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
475 functions will allow you to initiate a DUNDi query from the dialplan,
476 find out how many results there are, and access each one.
478 ENUM changes
479 ------------
480 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
481 functions will allow you to initiate an ENUM lookup from the dialplan,
482 and Asterisk will cache the results. ENUMRESULT can be used to access
483 the results without doing multiple DNS queries.
485 Voicemail Changes
486 -----------------
487 * Added the ability to customize which sound files are used for some of the
488 prompts within the Voicemail application by changing them in voicemail.conf
489 * Added the ability for the "voicemail show users" CLI command to show users
490 configured by the dynamic realtime configuration method.
491 * MWI (Message Waiting Indication) handling has been significantly
492 restructured internally to Asterisk. It is now totally event based
493 instead of polling based. The voicemail application will notify other
494 modules that have subscribed to MWI events when something in the mailbox
495 changes.
496 This also means that if any other entity outside of Asterisk is changing
497 the contents of mailboxes, then the voicemail application still needs to
498 poll for changes. Examples of situations that would require this option
499 are web interfaces to voicemail or an email client in the case of using
500 IMAP storage. So, two new options have been added to voicemail.conf
501 to account for this: "pollmailboxes" and "pollfreq". See the sample
502 configuration file for details.
503 * Added "tw" language support
504 * Added support for storage of greetings using an IMAP server
505 * Added ability to customize forward, reverse, stop, and pause keys for message playback
506 * SMDI is now enabled in voicemail using the smdienable option.
507 * A "lockmode" option has been added to asterisk.conf to configure the file
508 locking method used for voicemail, and potentially other things in the
509 future. The default is the old behavior, lockfile. However, there is a
510 new method, "flock", that uses a different method for situations where the
511 lockfile will not work, such as on SMB/CIFS mounts.
512 * Added the ability to backup deleted messages, to ease recovery in the case
513 that a user accidentally deletes a message, and discovers that they need it.
514 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
515 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
516 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
517 voicemail boxes. The SMDI interface can also poll for MWI changes when some
518 outside entity is modifying the state of the mailbox (such as IMAP storage or
519 a web interface of some kind).
520 * Added the support for marking messages as "urgent." There are two methods to accomplish
521 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
522 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
523 the message as urgent after he has recorded a voicemail by following the voice instructions.
524 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
525 messages
527 Queue changes
528 -------------
529 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
530 used across multiple queues.
531 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
532 setqueueentryvar options for each queue, see queues.conf.sample for details.
533 * Added keepstats option to queues.conf which will keep queue
534 statistics during a reload.
535 * setinterfacevar option in queues.conf also now sets a variable
536 called MEMBERNAME which contains the member's name.
537 * Added 'Strategy' field to manager event QueueParams which represents
538 the queue strategy in use.
539 * Added option to run macro when a queue member is connected to a caller,
540 see queues.conf.sample for details.
541 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
542 does not count paused queue members as unavailable.
543 * Added min-announce-frequency option to queues.conf which allows you to control the
544 minimum amount of time between queue announcements for use when the caller's queue
545 position changes frequently.
546 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
547 queue log.
548 * Added ability for non-realtime queues to have realtime members
549 * Added the "linear" strategy to queues.
550 * Added the "wrandom" strategy to queues.
551 * Added new channel variable QUEUE_MIN_PENALTY
552 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
553 rules in queuerules.conf. See configs/queuerules.conf.sample for details
554 * Added a new parameter for member definition, called state_interface. This may be
555 used so that a member may be called via one interface but have a different interface's
556 device state reported.
557 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
558 specified by the periodic-announce option, then one will be chosen randomly when it is time
559 to play a periodic announcment
560 * New configuration options: announce-position now takes two more values in addition to "yes" and
561 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
562 announce-position-limit. By setting announce-position to "limit" callers will only have their
563 position announced if their position is less than what is specified by announce-position-limit.
564 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
565 will be told that their are more than announce-position-limit callers waiting.
566 * Two new queue log events have been added. An ADDMEMBER event will be logged
567 when a realtime queue member is added and a REMOVEMEMBER event will be logged
568 when a realtime queue member is removed. Since there is no calling channel associated
569 with these events, the string "REALTIME" is placed where the channel's unique id
570 is typically placed.
572 MeetMe Changes
573 --------------
574 * The 'o' option to provide an optimization has been removed and its functionality
575 has been enabled by default.
576 * When a conference is created, the UNIQUEID of the channel that caused it to be
577 created is stored. Then, every channel that joins the conference will have the
578 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
579 callers that come and go from long standing conferences.
580 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
581 except it does operations on a channel by name, instead of number in a conference.
582 This is a very useful feature in combination with the 'X' option to ChanSpy.
583 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
584 when kicked out.
585 * Added new RealTime functionality to provide support for scheduled conferencing.
586 This includes optional messages to the caller if they attempt to join before
587 the schedule start time, or to allow the caller to join the conference early.
588 Also included is optional support for limiting the number of callers per
589 RealTime conference.
590 * Added the S() and L() options to the MeetMe application. These are pretty
591 much identical to the S() and L() options to Dial(). They let you set
592 timeouts for the conference, as well as have warning sounds played to
593 let the caller know how much time is left, and when it is running out.
594 * Added the ability to do "meetme concise" with the "meetme" CLI command.
595 This extends the concise capabilities of this CLI command to include
596 listing all conferences, instead of an addition to the other sub commands
597 for the "meetme" command.
598 * Added the ability to specify the music on hold class used to play into the
599 conference when there is only one member and the M option is used.
600 * Added MEETME_INFO dialplan function which provides a way to query
601 various properties of a Meetme conference.
603 Other Dialplan Application Changes
604 ----------------------------------
605 * Argument support for Gosub application
606 * From the to-do lists: straighten out the app timeout args:
607 Wait() app now really does 0.3 seconds- was truncating arg to an int.
608 WaitExten() same as Wait().
609 Congestion() - Now takes floating pt. argument.
610 Busy() - now takes floating pt. argument.
611 Read() - timeout now can be floating pt.
612 WaitForRing() now takes floating pt timeout arg.
613 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
614 * Added 's' option to Page application.
615 * Added 'E' and 'V' commands to ExternalIVR.
616 * Added 'o' and 'X' options to Chanspy.
617 * Added a new dialplan application, Bridge, which allows you to bridge the
618 calling channel to any other active channel on the system.
619 * Added the ability to specify a music on hold class to play instead of ringing
620 for the SLATrunk application.
621 * The Read application no longer exits the dialplan on error. Instead, it sets
622 READSTATUS to ERROR, which you can catch and handle separately.
623 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
624 of asking for verification of each name, one at a time.
625 * Privacy() no longer uses privacy.conf, as all options are specifyable as
626 direct options to the app.
627 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
628 for more details
629 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
630 * The ChannelRedirect application no longer exits the dialplan if the given channel
631 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
632 or NOCHANNEL if the given channel was not found.
633 * The silencethreshold setting that was previously configurable in multiple
634 applications is now settable globally via dsp.conf.
635 * Added ability to communicate over a TCP socket instead of forking a child process for the
636 ExternalIVR application.
638 Music On Hold Changes
639 ---------------------
640 * A new option, "digit", has been added for music on hold classes in
641 musiconhold.conf. If this is set for a music on hold class, a caller
642 listening to music on hold can press this digit to switch to listening
643 to this music on hold class.
644 * Support for realtime music on hold has been added.
645 * In conjunction with the realtime music on hold, a general section has
646 been added to musiconhold.conf, its sole variable is cachertclasses. If this
647 is set, then music on hold classes found in realtime will be cached in memory.
649 AEL Changes
650 -----------
651 * AEL upgraded to use the Gosub with Arguments instead
652 of Macro application, to hopefully reduce the problems
653 seen with the artificially low stack ceiling that
654 Macro bumps into. Macros can only call other Macros
655 to a depth of 7. Tests run using gosub, show depths
656 limited only by virtual memory. A small test demonstrated
657 recursive call depths of 100,000 without problems.
658 -- in addition to this, all apps that allowed a macro
659 to be called, as in Dial, queues, etc, are now allowing
660 a gosub call in similar fashion.
661 * AEL now generates LOCAL(argname) declarations when it
662 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
663 etc. That makes the arguments local in scope. The user
664 can define their own local variables in macros, now,
665 by saying "local myvar=someval;" or using Set() in this
666 fashion: Set(LOCAL(myvar)=someval); ("local" is now
667 an AEL keyword).
668 * utils/conf2ael introduced. Will convert an extensions.conf
669 file into extensions.ael. Very crude and unfinished, but
670 will be improved as time goes by. Should be useful for a
671 first pass at conversion.
672 * aelparse will now read extensions.conf to see if a referenced
673 macro or context is there before issueing a warning.
674 * AEL parser sets a local channel variable ~~EXTEN~~, to
675 preserve the value of ${EXTEN} thru switch statements.
676 * New operator in $[...] expressions: the ~~ operator serves
677 as a concatenation operator. AT THE MOMENT, it is really only
678 necessary and useful in AEL, especially in if() expressions.
679 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
680 any enclosing double-quotes, and evaluate to the value of a
681 concatenated with the value of b. For example if a is set to
682 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
683 evaluate to xyzabc .
686 Call Features (res_features) Changes
687 ------------------------------------
688 * Added the parkedcalltransfers option to features.conf
689 * The built-in method for doing attended transfers has been updated to
690 include some new options that allow you to have the transferee sent
691 back to the person that did the transfer if the transfer is not successful.
692 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
693 in features.conf.sample.
694 * Added support for configuring named groups of custom call features in
695 features.conf. This means that features can be written a single time, and
696 then mapped into groups of features for different key mappings or easier
697 access control.
698 * Updated the ParkedCall application to allow you to not specify a parking
699 extension. If you don't specify a parking space to pick up, it will grab
700 the first one available.
701 * Added cli command 'features reload' to reload call features from features.conf
702 * Moved into core asterisk binary.
704 Language Support Changes
705 ------------------------
706 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
707 * Added support for the Hungarian language for saying numbers, dates, and times.
709 AGI Changes
710 -----------
711 * Added SPEECH commands for speech recognition. A complete listing can be found
712 using agi show.
713 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
714 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
715 does not behave as expected; the native command needs to be used, instead.
717 Logger changes
718 --------------
719 * Added rotatestrategy option to logger.conf, along with two new options:
720 "timestamp" which will use the time to name the logger files instead of
721 sequence number; and "rotate", which rotates the names of the logfiles,
722 similar to the way syslog rotates files.
723 * Added exec_after_rotate option to logger.conf, which allows a system
724 command to be run after rotation. This is primarily useful with
725 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
726 and to ensure that the oldest log file gets deleted.
727 * Added realtime support for the queue log
729 Call Detail Records
730 -------------------
731 * The cdr_manager module has a [mappings] feature, like cdr_custom,
732 to add fields to the manager event from the CDR variables.
733 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
734 backend database CDR table. Specifically, additional, non-standard
735 columns are supported, merely by setting the corresponding CDR variable in
736 your dialplan. In addition, you may alias any column to another name (for
737 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
738 simply "alias src => ANI" in the configuration file). Records may be
739 posted to more than one backend, simply by specifying multiple categories
740 in the configuration file. And finally, you may filter which CDRs get
741 posted to each backend, by specifying a filter (which the record must
742 match) for the particular category. Filters are additive (meaning all
743 rules must match to post that CDR).
744 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
745 module. Specifically, you may add additional columns into the table and
746 they will be set, if you set the corresponding CDR variable name. Also,
747 if you omit columns in your database table, they will be silently skipped
748 (but a record will still be inserted, based on what columns remain). Note
749 that the other two features from cdr_adaptive_odbc (alias and filter) are
750 not currently supported.
751 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
752 has been disabled using the NoCDR application.
754 Miscellaneous New Modules
755 -------------------------
756 * Added a new CDR module, cdr_sqlite3_custom.
757 * Added a new realtime configuration module, res_config_sqlite
758 * Added a new codec translation module, codec_resample, which re-samples
759 signed linear audio between 8 kHz and 16 kHz to help support wideband
760 codecs.
761 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
762 based on configuration templates that use Asterisk dialplan function and
763 variable substitution. It should be possible to create phone profiles and
764 templates that work for the majority of phones provisioned over http. It
765 is currently only intended to provision a single user account per phone.
766 An example profile and set of templates for Polycom phones is provided.
767 NOTE: Polycom firmware is not included, but should be placed in
768 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
769 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
770 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
771 provided; there is a JACK() application, and a JACK_HOOK() function. Both
772 interfaces create an input and output JACK port. The application makes
773 these ports the endpoint of the call. The audio coming from the channel
774 goes out the output port and whatever comes back in on the input port is
775 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
776 audiohook on the channel. This lets you run the audio coming from a
777 channel through JACK, and whatever comes back in is what gets forwarded
778 on as the channel's audio. This is very useful for building custom
779 vocoders or doing recording or analysis of the channel's audio in another
780 application.
781 * Added a new module, res_config_curl, which permits using a HTTP POST url
782 to retrieve, create, update, and delete realtime information from a remote
783 web server. Note that this module requires func_curl.so to be loaded for
784 backend functionality.
785 * Added a new module, res_config_ldap, which permits the use of an LDAP
786 server for realtime data access.
787 * Added support for writing and running your dialplan in lua using the pbx_lua
788 module. See configs/extensions.lua.sample for examples of how to do this.
790 Miscellaneous
791 -------------
792 * Ability to use libcap to set high ToS bits when non-root
793 on Linux. If configure is unable to find libcap then you
794 can use --with-cap to specify the path.
795 * Added maxfiles option to options section of asterisk.conf which allows you to specify
796 what Asterisk should set as the maximum number of open files when it loads.
797 * Added the jittertargetextra configuration option.
798 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
799 configuration files for the IP channel drivers. The new option is "cos".
800 This information is also documented in doc/qos.tex, or the IP Quality of Service
801 section of asterisk.pdf.
802 * When originating a call using AMI or pbx_spool that fails the reason for failure
803 will now be available in the failed extension using the REASON dialplan variable.
804 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
805 It allows you to configure a prefix for auto-monitor recordings.
806 * A new extension pattern matching algorithm, based on a trie, is introduced
807 here, that could noticeably speed up mid-sized to large dialplans.
808 It is NOT used by default, as duplicating the behaviour of the old pattern
809 matcher is still under development. A config file option, in extensions.conf,
810 in the [general] section, called "extenpatternmatchingnew", is by default
811 set to false; setting that to true will force the use of the new algorithm.
812 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
813 be used to switch the algorithms at run time.
814 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
815 specifying which socket to use to connect to the running Asterisk daemon
816 (-s)
817 * Performance enhancements to the sched facility, which is used in
818 the channel drivers, etc. Added hashtabs and doubly-linked lists
819 to speed up deletion; start at the beginning or end of list to
820 speed up insertion.
821 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
822 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
823 Added regression tests to the tests/ dir, also.
824 * Added a refcount trace feature to astobj2 for those trying to balance
825 object creation, deletion; work, play; space and time. See the
826 notes in astobj2.h. Also, see utils/refcounter as well, as a
827 quick way to find unbalanced refcounts in what could be a sea
828 of objects that were balanced.
829 * Added logging to 'make update' command. See update.log
830 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
831 do not come from the remote party.
832 * Added the 'n' option to the SpeechBackground application to tell it to not
833 answer the channel if it has not already been answered.
834 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
835 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
836 dialplan debugging.
837 * iLBC source code no longer included (see UPGRADE.txt for details)
838 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
839 deadlock is detected, a backtrace of the stack which led to the lock calls
840 will be output to the CLI.
841 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
842 the "core show locks" CLI command will give lock information output as well
843 as a backtrace of the stack which led to the lock calls.
844 * users.conf now sports an optional alternateexts property, which permits
845 allocation of additional extensions which will reach the specified user.
846 * A new option for the configure script, --enable-internal-poll, has been added
847 for use with systems which may have a buggy implementation of the poll system
848 call. If you notice odd behavior such as the CLI being unresponsive on remote
849 consoles, you may want to try using this option. This option is enabled by default
850 on Darwin systems since it is known that the Darwin poll() implementation has
851 odd issues.