| blob | 0552d861b94a2bf5fe6ffb5f023466df59a3b637 |
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.
110 SIP Changes
111 -----------
112 * Added DNS manager support to registrations for peers referencing peer entries.
113 DNS manager runs in the background which allows DNS lookups to be run asynchronously
114 as well as periodically updating the IP address. These properties allow for
115 better performance as well as recovery in the event of an IP change.
116 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
117 load/reload of large numbers of peers/users by ~40x (for large lists of peers.
118 Initially, we saw 4x improvement in call setup/destruction, but at the time
119 of merging, this gain has disappeared; further research will be done to try
120 and restore this performance improvement. Astobj2 refcounting is now used
121 for users, peers, and dialogs. Users are encouraged to assist in regression
122 testing and problem reporting!
123 * Added ability to specify registration expiry time on a per registration basis in
124 the register line.
125 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
126 lost packets.
127 * Added t38pt_usertpsource option. See sip.conf.sample for details.
128 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
129 * 'sip show peers' and 'sip show users' display their entries sorted in
130 alphabetical order, as opposed to the order they were in, in the config
131 file or database.
132 * Videosupport now supports an additional option, "always", which always sets
133 up video RTP ports, even on clients that don't support it. This helps with
134 callfiles and certain transfers to ensure that if two video phones are
135 connected, they will always share video feeds.
137 IAX Changes
138 -----------
139 * Existing DNS manager lookups extended to check for SRV records.
141 CLI Changes
142 -----------
143 * New CLI command, "config reload <file.conf>" which reloads any module that
144 references that particular configuration file. Also added "config list"
145 which shows which configuration files are in use.
146 * New CLI commands, "pri show version" and "ss7 show version" that will
147 display which version of libpri and libss7 are being used, respectively.
148 A new API call was added so trunk will now have to be compiled against
149 a versions of libpri and libss7 that have them or it will not know that
150 these libraries exist.
151 * The commands "core show globals", "core set global" and "core set chanvar" has
152 been deprecated in favor of the more semanticly correct "dialplan show globals",
153 "dialplan set chanvar" and "dialplan set global".
154 * New CLI command "dialplan show chanvar" to list all variables associated
155 with a given channel.
157 DNS manager changes
158 -------------------
159 * Addresses managed by DNS manager now can check to see if there is a DNS
160 SRV record for a given domain and will use that hostname/port if present.
162 AMI - The manager (TCP/TLS/HTTP)
163 --------------------------------
164 * The Status command now takes an optional list of variables to display
165 along with channel status.
167 ODBC Changes
168 ------------
169 * res_odbc no longer has a limit of 1023 total possible unshared connections,
170 as some people were running into this limit. This limit has been increased
171 to 4.2 billion.
173 Queue changes
174 -------------
175 * The TRANSFER queue log entry now includes the the caller's original
176 position in the transferred-from queue.
177 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
178 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
179 as well as an explanation about timeout options in general
181 Realtime changes
182 ----------------
183 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
184 adaptive capabilities. What this means in practical terms is that if your
185 realtime table lacks critical fields, Asterisk will now emit warnings to
186 that effect. Also, some of the realtime drivers have the ability (if
187 configured) to automatically add those columns to the table with the
188 correct type and length.
190 Miscellaneous
191 -------------
192 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
193 the 'setvar' option to cause a given audio file to be played upon completion
194 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
195 Skinny channels only.
197 ------------------------------------------------------------------------------
198 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
199 ------------------------------------------------------------------------------
201 AMI - The manager (TCP/TLS/HTTP)
202 --------------------------------
203 * Manager has undergone a lot of changes, all of them documented
204 in doc/manager_1_1.txt
205 * Manager version has changed to 1.1
206 * Added a new action 'CoreShowChannels' to list currently defined channels
207 and some information about them.
208 * Added a new action 'SIPshowregistry' to list SIP registrations.
209 * Added TLS support for the manager interface and HTTP server
210 * Added the URI redirect option for the built-in HTTP server
211 * The output of CallerID in Manager events is now more consistent.
212 CallerIDNum is used for number and CallerIDName for name.
213 * Enable https support for builtin web server.
214 See configs/http.conf.sample for details.
215 * Added a new action, GetConfigJSON, which can return the contents of an
216 Asterisk configuration file in JSON format. This is intended to help
217 improve the performance of AJAX applications using the manager interface
218 over HTTP.
219 * SIP and IAX manager events now use "ChannelType" in all cases where we
220 indicate channel driver. Previously, we used a mixture of "Channel"
221 and "ChannelDriver" headers.
222 * Added a "Bridge" action which allows you to bridge any two channels that
223 are currently active on the system.
224 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
225 the voicemail users setup.
226 * Added 'DBDel' and 'DBDelTree' manager commands.
227 * cdr_manager now reports events via the "cdr" level, separating it from
228 the very verbose "call" level.
229 * Manager users are now stored in memory. If you change the manager account
230 list (delete or add accounts) you need to reload manager.
231 * Added Masquerade manager event for when a masquerade happens between
232 two channels.
233 * Added "manager reload" command for the CLI
234 * Lots of commands that only provided information are now allowed under the
235 Reporting privilege, instead of only under Call or System.
236 * The IAX* commands now require either System or Reporting privilege, to
237 mirror the privileges of the SIP* commands.
238 * Added ability to retrieve list of categories in a config file.
239 * Added ability to retrieve the content of a particular category.
240 * Added ability to empty a context.
241 * Created new action to create a new file.
242 * Updated delete action to allow deletion by line number with respect to category.
243 * Added new action insert to add new variable to category at specified line.
244 * Updated action newcat to allow new category to be inserted in file above another
245 existing category.
246 * Added new event "JitterBufStats" in the IAX2 channel
247 * Originate now requires the Originate privilege and, if you want to call out
248 to a subshell, it requires the System privilege, as well. This was done to
249 enhance manager security.
250 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
251 * New command: Atxfer. See doc/manager_1_1.txt for more details or
252 manager show command Atxfer from the CLI
254 Dialplan functions
255 ------------------
256 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
257 state in the dialplan, as well as creating custom device states that are
258 controllable from the dialplan.
259 * Extend CALLERID() function with "pres" and "ton" parameters to
260 fetch string representation of calling number presentation indicator
261 and numeric representation of type of calling number value.
262 * MailboxExists converted to dialplan function
263 * A new option to Dial() for telling IP phones not to count the call
264 as "missed" when dial times out and cancels.
265 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
266 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
267 held for any given channel. Also, locks are automatically freed when a
268 channel is hung up.
269 * Added HINT() dialplan function that allows retrieving hint information.
270 Hints are mappings between extensions and devices for the sake of
271 determining the state of an extension. This function can retrieve the list
272 of devices or the name associated with a hint.
273 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
274 of any extension.
275 * Added SYSINFO() dialplan function which allows retrieval of system information
276 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
277 the existence of a dialplan target.
278 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
279 upper and lower case, respectively.
280 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
281 ID for the call (not the Asterisk call ID or unique ID), provided that the
282 channel driver supports this. For SIP, you get the SIP call-ID for the
283 bridged channel which you can store in the CDR with a custom field.
285 CLI Changes
286 -----------
287 * New CLI command "core show hint" (usage: core show hint <exten>)
288 * New CLI command "core show settings"
289 * Added 'core show channels count' CLI command.
290 * Added the ability to set the core debug and verbose values on a per-file basis.
291 * Added 'queue pause member' and 'queue unpause member' CLI commands
292 * Ability to set process limits ("ulimit") without restarting Asterisk
293 * Enhanced "agi debug" to print the channel name as a prefix to the debug
294 output to make debugging on busy systems much easier.
295 * New CLI commands "dialplan set extenpatternmatching true/false"
296 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
297 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
298 listed in the startup_commands section of cli.conf will get executed.
299 * Added a CLI command, "devstate change", which allows you to set custom device
300 states from the func_devstate module that provides the DEVICE_STATE() function
301 and handling of the "Custom:" devices.
302 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
303 sorted into the different possible callbacks, with the number of entries
304 currently scheduled for each. Gives you a feel for how busy the sip channel
305 driver is.
307 SIP changes
308 -----------
309 * Improved NAT and STUN support.
310 chan_sip now can use port numbers in bindaddr, externip and externhost
311 options, as well as contact a STUN server to detect its external address
312 for the SIP socket. See sip.conf.sample, 'NAT' section.
313 * The default SIP useragent= identifier now includes the Asterisk version
314 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
315 If set, and the incoming request carries authentication info,
316 the username to match in the users list is taken from the Digest header
317 rather than from the From: field. This feature is considered experimental.
318 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
319 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
320 * The "localmask" setting was removed in version 1.2 and the reminder about it
321 being removed is now also removed.
322 * A new option "busylevel" for setting a level of calls where asterisk reports
323 a device as busy, to separate it from call-limit. This value is also added
324 to the SIP_PEER dialplan function.
325 * A new realtime family called "sipregs" is now supported to store SIP registration
326 data. If this family is defined, "sippeers" will be used for configuration and
327 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
328 registration data, as before.
329 * The SIPPEER function have new options for port address, call and pickup groups
330 * Added support for T.140 realtime text in SIP/RTP
331 * The "checkmwi" option has been removed from sip.conf, as it is no longer
332 required due to the restructuring of how MWI is handled. See the descriptions
333 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
334 for more information.
335 * Added rtpdest option to CHANNEL() dialplan function.
336 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
337 * SIP now adds a header to the CANCEL if the call was answered by another phone
338 in the same dial command, or if the new c option in dial() is used.
339 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
340 states it is not needed. For phones, however, that do require it the "registertrying" option
341 has been added so it can be enabled.
342 * A new option called "callcounter" (global/peer/user level) enables call counters needed
343 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
344 used to enable this functionality).
345 * New settings for timer T1 and timer B on a global level or per device. This makes it
346 possible to force timeout faster on non-responsive SIP servers. These settings are
347 considered advanced, so don't use them unless you have a problem.
348 * Added a dial string option to be able to set the To: header in an INVITE to any
349 SIP uri.
350 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
351 the qualify frequency.
352 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
353 were not properly torn down due to network or endpoint failures during an established
354 SIP session.
355 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
356 configs/sip.conf.sample for more information on how it is used.
357 * Added a new configuration option "authfailureevents" that enables manager events when
358 a peer can't authenticate properly.
359 * Added DNS manager support to registrations for peers not referencing a peer entry.
361 IAX2 changes
362 ------------
363 * Added the trunkmaxsize configuration option to chan_iax2.
364 * Added the srvlookup option to iax.conf
365 * Added support for OSP. The token is set and retrieved through the CHANNEL()
366 dialplan function.
368 XMPP Google Talk/Jingle changes
369 -------------------------------
370 * Added the bindaddr option to gtalk.conf.
372 Skinny changes
373 -------------
374 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
375 * Proper codec support in chan_skinny.
376 * Added settings for IP and Ethernet QoS requests
378 MGCP changes
379 ------------
380 * Added separate settings for media QoS in mgcp.conf
382 Console Channel Driver changes
383 ------------------------------
384 * Added experimental support for video send & receive to chan_oss.
385 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
386 a video source.
388 Phone channel changes (chan_phone)
389 ----------------------------------
390 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
392 H.323 channel Changes
393 ---------------------
394 * H323 remote hold notification support added (by NOTIFY message
395 and/or H.450 supplementary service)
397 Local channel changes
398 ---------------------
399 * The device state functionality in the Local channel driver has been updated
400 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
401 to just UNKNOWN if the extension exists.
402 * Added jitterbuffer support for chan_local. This allows you to use the
403 generic jitterbuffer on incoming calls going to Asterisk applications.
404 For example, this would allow you to use a jitterbuffer for an incoming
405 SIP call to Voicemail by putting a Local channel in the middle. This
406 feature is enabled by using the 'j' option in the Dial string to the Local
407 channel in conjunction with the existing 'n' option for local channels.
408 * A 'b' option has been added which causes chan_local to return the actual channel
409 that is behind it when queried. This is useful for transfer scenarios as the
410 actual channel will be transferred, not the Local channel.
412 Agent channel changes
413 ----------------------
414 * The ackcall and endcall options are now supplemented with options acceptdtmf
415 and enddtmf. These allow for the DTMF keypress to be configurable. The options
416 default to their old hard-coded values ('#' and '*' respectively) so this should
417 not break any existing agent installations.
419 DAHDI channel driver (chan_dahdi) Changes
420 ----------------------------------------
421 * SS7 support (via libss7 library)
422 * In India, some carriers transmit CID via dtmf. Some code has been added
423 that will handle some situations. The cidstart=polarity_IN choice has been added for
424 those carriers that transmit CID via dtmf after a polarity change.
425 * CID matching information is now shown when doing 'dialplan show'.
426 * Added dahdi show version CLI command.
427 * Added setvar support to chan_dahdi.conf channel entries.
428 * Added two new options: mwimonitor and mwimonitornotify. These options allow
429 you to enable MWI monitoring on FXO lines. When the MWI state changes,
430 the script specified in the mwimonitornotify option is executed. An internal
431 event indicating the new state of the mailbox is also generated, so that
432 the normal MWI facilities in Asterisk work as usual.
433 * Added signalling type 'auto', which attempts to use the same signalling type
434 for a channel as configured in DAHDI. This is primarily designed for analog
435 ports, but will also work for digital ports that are configured for FXS or FXO
436 signalling types. This mode is also the default now, so if your chan_dahdi.conf
437 does not specify signalling for a channel (which is unlikely as the sample
438 configuration file has always recommended specifying it for every channel) then
439 the 'auto' mode will be used for that channel if possible.
440 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
441 state for a channel; also ensured that the DNDState Manager event is
442 emitted no matter how the DND state is set or cleared.
444 New Channel Drivers
445 -------------------
446 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
447 configs/unistim.conf.sample for details. This new channel driver allows
448 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
449 * Added a new channel driver, chan_console, which uses portaudio as a cross
450 platform audio interface. It was written as a channel driver that would
451 work with Mac CoreAudio, but portaudio supports a number of other audio
452 interfaces, as well. Note that this channel driver requires v19 or higher
453 of portaudio; older versions have a different API.
455 DUNDi changes
456 -------------
457 * Added the ability to specify arguments to the Dial application when using
458 the DUNDi switch in the dialplan.
459 * Added the ability to set weights for responses dynamically. This can be
460 done using a global variable or a dialplan function. Using the SHELL()
461 function would allow you to have an external script set the weight for
462 each response.
463 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
464 functions will allow you to initiate a DUNDi query from the dialplan,
465 find out how many results there are, and access each one.
467 ENUM changes
468 ------------
469 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
470 functions will allow you to initiate an ENUM lookup from the dialplan,
471 and Asterisk will cache the results. ENUMRESULT can be used to access
472 the results without doing multiple DNS queries.
474 Voicemail Changes
475 -----------------
476 * Added the ability to customize which sound files are used for some of the
477 prompts within the Voicemail application by changing them in voicemail.conf
478 * Added the ability for the "voicemail show users" CLI command to show users
479 configured by the dynamic realtime configuration method.
480 * MWI (Message Waiting Indication) handling has been significantly
481 restructured internally to Asterisk. It is now totally event based
482 instead of polling based. The voicemail application will notify other
483 modules that have subscribed to MWI events when something in the mailbox
484 changes.
485 This also means that if any other entity outside of Asterisk is changing
486 the contents of mailboxes, then the voicemail application still needs to
487 poll for changes. Examples of situations that would require this option
488 are web interfaces to voicemail or an email client in the case of using
489 IMAP storage. So, two new options have been added to voicemail.conf
490 to account for this: "pollmailboxes" and "pollfreq". See the sample
491 configuration file for details.
492 * Added "tw" language support
493 * Added support for storage of greetings using an IMAP server
494 * Added ability to customize forward, reverse, stop, and pause keys for message playback
495 * SMDI is now enabled in voicemail using the smdienable option.
496 * A "lockmode" option has been added to asterisk.conf to configure the file
497 locking method used for voicemail, and potentially other things in the
498 future. The default is the old behavior, lockfile. However, there is a
499 new method, "flock", that uses a different method for situations where the
500 lockfile will not work, such as on SMB/CIFS mounts.
501 * Added the ability to backup deleted messages, to ease recovery in the case
502 that a user accidentally deletes a message, and discovers that they need it.
503 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
504 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
505 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
506 voicemail boxes. The SMDI interface can also poll for MWI changes when some
507 outside entity is modifying the state of the mailbox (such as IMAP storage or
508 a web interface of some kind).
509 * Added the support for marking messages as "urgent." There are two methods to accomplish
510 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
511 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
512 the message as urgent after he has recorded a voicemail by following the voice instructions.
513 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
514 messages
516 Queue changes
517 -------------
518 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
519 used across multiple queues.
520 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
521 setqueueentryvar options for each queue, see queues.conf.sample for details.
522 * Added keepstats option to queues.conf which will keep queue
523 statistics during a reload.
524 * setinterfacevar option in queues.conf also now sets a variable
525 called MEMBERNAME which contains the member's name.
526 * Added 'Strategy' field to manager event QueueParams which represents
527 the queue strategy in use.
528 * Added option to run macro when a queue member is connected to a caller,
529 see queues.conf.sample for details.
530 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
531 does not count paused queue members as unavailable.
532 * Added min-announce-frequency option to queues.conf which allows you to control the
533 minimum amount of time between queue announcements for use when the caller's queue
534 position changes frequently.
535 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
536 queue log.
537 * Added ability for non-realtime queues to have realtime members
538 * Added the "linear" strategy to queues.
539 * Added the "wrandom" strategy to queues.
540 * Added new channel variable QUEUE_MIN_PENALTY
541 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
542 rules in queuerules.conf. See configs/queuerules.conf.sample for details
543 * Added a new parameter for member definition, called state_interface. This may be
544 used so that a member may be called via one interface but have a different interface's
545 device state reported.
546 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
547 specified by the periodic-announce option, then one will be chosen randomly when it is time
548 to play a periodic announcment
549 * New configuration options: announce-position now takes two more values in addition to "yes" and
550 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
551 announce-position-limit. By setting announce-position to "limit" callers will only have their
552 position announced if their position is less than what is specified by announce-position-limit.
553 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
554 will be told that their are more than announce-position-limit callers waiting.
555 * Two new queue log events have been added. An ADDMEMBER event will be logged
556 when a realtime queue member is added and a REMOVEMEMBER event will be logged
557 when a realtime queue member is removed. Since there is no calling channel associated
558 with these events, the string "REALTIME" is placed where the channel's unique id
559 is typically placed.
561 MeetMe Changes
562 --------------
563 * The 'o' option to provide an optimization has been removed and its functionality
564 has been enabled by default.
565 * When a conference is created, the UNIQUEID of the channel that caused it to be
566 created is stored. Then, every channel that joins the conference will have the
567 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
568 callers that come and go from long standing conferences.
569 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
570 except it does operations on a channel by name, instead of number in a conference.
571 This is a very useful feature in combination with the 'X' option to ChanSpy.
572 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
573 when kicked out.
574 * Added new RealTime functionality to provide support for scheduled conferencing.
575 This includes optional messages to the caller if they attempt to join before
576 the schedule start time, or to allow the caller to join the conference early.
577 Also included is optional support for limiting the number of callers per
578 RealTime conference.
579 * Added the S() and L() options to the MeetMe application. These are pretty
580 much identical to the S() and L() options to Dial(). They let you set
581 timeouts for the conference, as well as have warning sounds played to
582 let the caller know how much time is left, and when it is running out.
583 * Added the ability to do "meetme concise" with the "meetme" CLI command.
584 This extends the concise capabilities of this CLI command to include
585 listing all conferences, instead of an addition to the other sub commands
586 for the "meetme" command.
587 * Added the ability to specify the music on hold class used to play into the
588 conference when there is only one member and the M option is used.
589 * Added MEETME_INFO dialplan function which provides a way to query
590 various properties of a Meetme conference.
592 Other Dialplan Application Changes
593 ----------------------------------
594 * Argument support for Gosub application
595 * From the to-do lists: straighten out the app timeout args:
596 Wait() app now really does 0.3 seconds- was truncating arg to an int.
597 WaitExten() same as Wait().
598 Congestion() - Now takes floating pt. argument.
599 Busy() - now takes floating pt. argument.
600 Read() - timeout now can be floating pt.
601 WaitForRing() now takes floating pt timeout arg.
602 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
603 * Added 's' option to Page application.
604 * Added 'E' and 'V' commands to ExternalIVR.
605 * Added 'o' and 'X' options to Chanspy.
606 * Added a new dialplan application, Bridge, which allows you to bridge the
607 calling channel to any other active channel on the system.
608 * Added the ability to specify a music on hold class to play instead of ringing
609 for the SLATrunk application.
610 * The Read application no longer exits the dialplan on error. Instead, it sets
611 READSTATUS to ERROR, which you can catch and handle separately.
612 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
613 of asking for verification of each name, one at a time.
614 * Privacy() no longer uses privacy.conf, as all options are specifyable as
615 direct options to the app.
616 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
617 for more details
618 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
619 * The ChannelRedirect application no longer exits the dialplan if the given channel
620 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
621 or NOCHANNEL if the given channel was not found.
622 * The silencethreshold setting that was previously configurable in multiple
623 applications is now settable globally via dsp.conf.
624 * Added ability to communicate over a TCP socket instead of forking a child process for the
625 ExternalIVR application.
627 Music On Hold Changes
628 ---------------------
629 * A new option, "digit", has been added for music on hold classes in
630 musiconhold.conf. If this is set for a music on hold class, a caller
631 listening to music on hold can press this digit to switch to listening
632 to this music on hold class.
633 * Support for realtime music on hold has been added.
634 * In conjunction with the realtime music on hold, a general section has
635 been added to musiconhold.conf, its sole variable is cachertclasses. If this
636 is set, then music on hold classes found in realtime will be cached in memory.
638 AEL Changes
639 -----------
640 * AEL upgraded to use the Gosub with Arguments instead
641 of Macro application, to hopefully reduce the problems
642 seen with the artificially low stack ceiling that
643 Macro bumps into. Macros can only call other Macros
644 to a depth of 7. Tests run using gosub, show depths
645 limited only by virtual memory. A small test demonstrated
646 recursive call depths of 100,000 without problems.
647 -- in addition to this, all apps that allowed a macro
648 to be called, as in Dial, queues, etc, are now allowing
649 a gosub call in similar fashion.
650 * AEL now generates LOCAL(argname) declarations when it
651 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
652 etc. That makes the arguments local in scope. The user
653 can define their own local variables in macros, now,
654 by saying "local myvar=someval;" or using Set() in this
655 fashion: Set(LOCAL(myvar)=someval); ("local" is now
656 an AEL keyword).
657 * utils/conf2ael introduced. Will convert an extensions.conf
658 file into extensions.ael. Very crude and unfinished, but
659 will be improved as time goes by. Should be useful for a
660 first pass at conversion.
661 * aelparse will now read extensions.conf to see if a referenced
662 macro or context is there before issueing a warning.
663 * AEL parser sets a local channel variable ~~EXTEN~~, to
664 preserve the value of ${EXTEN} thru switch statements.
665 * New operator in $[...] expressions: the ~~ operator serves
666 as a concatenation operator. AT THE MOMENT, it is really only
667 necessary and useful in AEL, especially in if() expressions.
668 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
669 any enclosing double-quotes, and evaluate to the value of a
670 concatenated with the value of b. For example if a is set to
671 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
672 evaluate to xyzabc .
675 Call Features (res_features) Changes
676 ------------------------------------
677 * Added the parkedcalltransfers option to features.conf
678 * The built-in method for doing attended transfers has been updated to
679 include some new options that allow you to have the transferee sent
680 back to the person that did the transfer if the transfer is not successful.
681 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
682 in features.conf.sample.
683 * Added support for configuring named groups of custom call features in
684 features.conf. This means that features can be written a single time, and
685 then mapped into groups of features for different key mappings or easier
686 access control.
687 * Updated the ParkedCall application to allow you to not specify a parking
688 extension. If you don't specify a parking space to pick up, it will grab
689 the first one available.
690 * Added cli command 'features reload' to reload call features from features.conf
691 * Moved into core asterisk binary.
693 Language Support Changes
694 ------------------------
695 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
696 * Added support for the Hungarian language for saying numbers, dates, and times.
698 AGI Changes
699 -----------
700 * Added SPEECH commands for speech recognition. A complete listing can be found
701 using agi show.
702 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
703 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
704 does not behave as expected; the native command needs to be used, instead.
706 Logger changes
707 --------------
708 * Added rotatestrategy option to logger.conf, along with two new options:
709 "timestamp" which will use the time to name the logger files instead of
710 sequence number; and "rotate", which rotates the names of the logfiles,
711 similar to the way syslog rotates files.
712 * Added exec_after_rotate option to logger.conf, which allows a system
713 command to be run after rotation. This is primarily useful with
714 rotatestrategry=rotate, to allow a limit on the number of logfiles kept
715 and to ensure that the oldest log file gets deleted.
716 * Added realtime support for the queue log
718 Call Detail Records
719 -------------------
720 * The cdr_manager module has a [mappings] feature, like cdr_custom,
721 to add fields to the manager event from the CDR variables.
722 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
723 backend database CDR table. Specifically, additional, non-standard
724 columns are supported, merely by setting the corresponding CDR variable in
725 your dialplan. In addition, you may alias any column to another name (for
726 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
727 simply "alias src => ANI" in the configuration file). Records may be
728 posted to more than one backend, simply by specifying multiple categories
729 in the configuration file. And finally, you may filter which CDRs get
730 posted to each backend, by specifying a filter (which the record must
731 match) for the particular category. Filters are additive (meaning all
732 rules must match to post that CDR).
733 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
734 module. Specifically, you may add additional columns into the table and
735 they will be set, if you set the corresponding CDR variable name. Also,
736 if you omit columns in your database table, they will be silently skipped
737 (but a record will still be inserted, based on what columns remain). Note
738 that the other two features from cdr_adaptive_odbc (alias and filter) are
739 not currently supported.
740 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
741 has been disabled using the NoCDR application.
743 Miscellaneous New Modules
744 -------------------------
745 * Added a new CDR module, cdr_sqlite3_custom.
746 * Added a new realtime configuration module, res_config_sqlite
747 * Added a new codec translation module, codec_resample, which re-samples
748 signed linear audio between 8 kHz and 16 kHz to help support wideband
749 codecs.
750 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
751 based on configuration templates that use Asterisk dialplan function and
752 variable substitution. It should be possible to create phone profiles and
753 templates that work for the majority of phones provisioned over http. It
754 is currently only intended to provision a single user account per phone.
755 An example profile and set of templates for Polycom phones is provided.
756 NOTE: Polycom firmware is not included, but should be placed in
757 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
758 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
759 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
760 provided; there is a JACK() application, and a JACK_HOOK() function. Both
761 interfaces create an input and output JACK port. The application makes
762 these ports the endpoint of the call. The audio coming from the channel
763 goes out the output port and whatever comes back in on the input port is
764 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
765 audiohook on the channel. This lets you run the audio coming from a
766 channel through JACK, and whatever comes back in is what gets forwarded
767 on as the channel's audio. This is very useful for building custom
768 vocoders or doing recording or analysis of the channel's audio in another
769 application.
770 * Added a new module, res_config_curl, which permits using a HTTP POST url
771 to retrieve, create, update, and delete realtime information from a remote
772 web server. Note that this module requires func_curl.so to be loaded for
773 backend functionality.
774 * Added a new module, res_config_ldap, which permits the use of an LDAP
775 server for realtime data access.
776 * Added support for writing and running your dialplan in lua using the pbx_lua
777 module. See configs/extensions.lua.sample for examples of how to do this.
779 Miscellaneous
780 -------------
781 * Ability to use libcap to set high ToS bits when non-root
782 on Linux. If configure is unable to find libcap then you
783 can use --with-cap to specify the path.
784 * Added maxfiles option to options section of asterisk.conf which allows you to specify
785 what Asterisk should set as the maximum number of open files when it loads.
786 * Added the jittertargetextra configuration option.
787 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
788 configuration files for the IP channel drivers. The new option is "cos".
789 This information is also documented in doc/qos.tex, or the IP Quality of Service
790 section of asterisk.pdf.
791 * When originating a call using AMI or pbx_spool that fails the reason for failure
792 will now be available in the failed extension using the REASON dialplan variable.
793 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
794 It allows you to configure a prefix for auto-monitor recordings.
795 * A new extension pattern matching algorithm, based on a trie, is introduced
796 here, that could noticeably speed up mid-sized to large dialplans.
797 It is NOT used by default, as duplicating the behaviour of the old pattern
798 matcher is still under development. A config file option, in extensions.conf,
799 in the [general] section, called "extenpatternmatchingnew", is by default
800 set to false; setting that to true will force the use of the new algorithm.
801 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
802 be used to switch the algorithms at run time.
803 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
804 specifying which socket to use to connect to the running Asterisk daemon
805 (-s)
806 * Performance enhancements to the sched facility, which is used in
807 the channel drivers, etc. Added hashtabs and doubly-linked lists
808 to speed up deletion; start at the beginning or end of list to
809 speed up insertion.
810 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
811 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
812 Added regression tests to the tests/ dir, also.
813 * Added a refcount trace feature to astobj2 for those trying to balance
814 object creation, deletion; work, play; space and time. See the
815 notes in astobj2.h. Also, see utils/refcounter as well, as a
816 quick way to find unbalanced refcounts in what could be a sea
817 of objects that were balanced.
818 * Added logging to 'make update' command. See update.log
819 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
820 do not come from the remote party.
821 * Added the 'n' option to the SpeechBackground application to tell it to not
822 answer the channel if it has not already been answered.
823 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
824 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
825 dialplan debugging.
826 * iLBC source code no longer included (see UPGRADE.txt for details)
827 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
828 deadlock is detected, a backtrace of the stack which led to the lock calls
829 will be output to the CLI.
830 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
831 the "core show locks" CLI command will give lock information output as well
832 as a backtrace of the stack which led to the lock calls.
833 * users.conf now sports an optional alternateexts property, which permits
834 allocation of additional extensions which will reach the specified user.
835 * A new option for the configure script, --enable-internal-poll, has been added
836 for use with systems which may have a buggy implementation of the poll system
837 call. If you notice odd behavior such as the CLI being unresponsive on remote
838 consoles, you may want to try using this option. This option is enabled by default
839 on Darwin systems since it is known that the Darwin poll() implementation has
840 odd issues.