1 -------------------------------------------------------------------------------
2 --- Functionality changes since Asterisk 1.4-beta was branched ----------------
3 -------------------------------------------------------------------------------
5 AMI - The manager (TCP/TLS/HTTP)
6 --------------------------------
7 * Added TLS support for the manager interface and HTTP server
8 * Added the URI redirect option for the built-in HTTP server
9 * The output of CallerID in Manager events is now more consistent.
10 CallerIDNum is used for number and CallerIDName for name.
11 * enable https support for builtin web server.
12 See configs/http.conf.sample for details.
13 * Added a new action, GetConfigJSON, which can return the contents of an
14 Asterisk configuration file in JSON format. This is intended to help
15 improve the performance of AJAX applications using the manager interface
17 * SIP and IAX manager events now use "ChannelType" in all cases where we
18 indicate channel driver. Previously, we used a mixture of "Channel"
19 and "ChannelDriver" headers.
20 * Added a "Bridge" action which allows you to bridge any two channels that
21 are currently active on the system.
22 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
23 the voicemail users setup.
24 * Added 'DBDel' and 'DBDelTree' manager commands.
28 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
29 state in the dialplan, as well as creating custom device states that are
30 controllable from the dialplan.
31 * Extend CALLERID() function with "pres" and "ton" parameters to
32 fetch string representation of calling number presentation indicator
33 and numeric representation of type of calling number value.
34 * MailboxExists converted to dialplan function
35 * A new option to Dial() for telling IP phones not to count the call
36 as "missed" when dial times out and cancels.
37 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
38 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
39 held for any given channel. Also, locks are automatically freed when a
41 * Added HINT() dialplan function that allows retrieving hint information.
42 Hints are mappings between extensions and devices for the sake of
43 determining the state of an extension. This function can retrieve the list
44 of devices or the name associated with a hint.
45 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
50 * New CLI command "core show settings"
51 * Added 'core show channels count' CLI command.
52 * Added the ability to set the core debug and verbose values on a per-file basis.
56 * Improved NAT and STUN support.
57 chan_sip now can use port numbers in bindaddr, externip and externhost
58 options, as well as contact a STUN server to detect its external address
59 for the SIP socket. See sip.conf.sample, 'NAT' section.
60 * The default SIP useragent= identifier now includes the Asterisk version
61 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
62 If set, and the incoming request carries authentication info,
63 the username to match in the users list is taken from the Digest header
64 rather than from the From: field. This feature is considered experimental.
65 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
66 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
67 * The "localmask" setting was removed in version 1.2 and the reminder about it
68 being removed is now also removed.
69 * A new option "busy-level" for setting a level of calls where asterisk reports
70 a device as busy, to separate it from call-limit
71 * A new realtime family called "sipregs" is now supported to store SIP registration
72 data. If this family is defined, "sippeers" will be used for configuration and
73 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
74 registration data, as before.
75 * The SIPPEER function have new options for port address, call and pickup groups
76 * Added support for T.140 realtime text in SIP/RTP
77 * The "checkmwi" option has been removed from sip.conf, as it is no longer
78 required due to the restructuring of how MWI is handled. See the descriptions
79 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
81 * Added rtpdest option to CHANNEL() dialplan function.
82 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
83 * SIP now adds a header to the CANCEL if the call was answered by another phone
84 in the same dial command, or if the new c option in dial() is used.
85 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
86 states it is not needed. For phones, however, that do require it the registertrying option
87 has been added so it can be enabled.
91 * Added the trunkmaxsize configuration option to chan_iax2.
92 * Added the srvlookup option to iax.conf
93 * Added support for OSP. The token is set and retrieved through the CHANNEL()
98 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
102 * Added the ability to specify arguments to the Dial application when using
103 the DUNDi switch in the dialplan.
104 * Added the ability to set weights for responses dynamically. This can be
105 done using a global variable or a dialplan function. Using the SHELL()
106 function would allow you to have an external script set the weight for
108 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
109 functions will allow you to initiate a DUNDi query from the dialplan,
110 find out how many results there are, and access each one.
114 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
115 functions will allow you to initiate an ENUM lookup from the dialplan,
116 and Asterisk will cache the results. ENUMRESULT can be used to access
117 the results without doing multiple DNS queries.
121 * Added the ability to customize which sound files are used for some of the
122 prompts within the Voicemail application by changing them in voicemail.conf
123 * Added the ability for the "voicemail show users" CLI command to show users
124 configured by the dynamic realtime configuration method.
125 * MWI (Message Waiting Indication) handling has been significantly
126 restructured internally to Asterisk. It is now totally event based
127 instead of polling based. The voicemail application will notify other
128 modules that have subscribed to MWI events when something in the mailbox
130 This also means that if any other entity outside of Asterisk is changing
131 the contents of mailboxes, then the voicemail application still needs to
132 poll for changes. Examples of situations that would require this option
133 are web interfaces to voicemail or an email client in the case of using
134 IMAP storage. So, two new options have been added to voicemail.conf
135 to account for this: "pollmailboxes" and "pollfreq". See the sample
136 configuration file for details.
137 * Added "tw" language support
138 * Added support for storage of greetings using an IMAP server
139 * Added ability to customize forward, reverse, stop, and pause keys for message playback
140 * SMDI is now enabled in voicemail using the smdienable option.
141 * A "lockmode" option has been added to asterisk.conf to configure the file
142 locking method used for voicemail, and potentially other things in the
143 future. The default is the old behavior, lockfile. However, there is a
144 new method, "flock", that uses a different method for situations where the
145 lockfile will not work, such as on SMB/CIFS mounts.
149 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
150 setqueueentryvar options for each queue, see queues.conf.sample for details.
151 * Added keepstats option to queues.conf which will keep queue
152 statistics during a reload.
153 * setinterfacevar option in queues.conf also now sets a variable
154 called MEMBERNAME which contains the member's name.
155 * Added 'Strategy' field to manager event QueueParams which represents
156 the queue strategy in use.
157 * Added option to run macro when a queue member is connected to a caller,
158 see queues.conf.sample for details.
159 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
160 does not count paused queue members as unavailable.
161 * Added min-announce-frequency option to queues.conf which allows you to control the
162 minimum amount of time between queue announcements for use when the caller's queue
163 position changes frequently.
164 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
166 * Added ability for non-realtime queues to have realtime members
170 * The 'o' option to provide an optimization has been removed and its functionality
171 has been enabled by default.
172 * When a conference is created, the UNIQUEID of the channel that caused it to be
173 created is stored. Then, every channel that joins the conference will have the
174 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
175 callers that come and go from long standing conferences.
176 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
177 except it does operations on a channel by name, instead of number in a conference.
178 This is a very useful feature in combination with the 'X' option to ChanSpy.
179 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
182 Music On Hold Changes
183 ---------------------
184 * A new option, "digit", has been added for music on hold classes in
185 musiconhold.conf. If this is set for a music on hold class, a caller
186 listening to music on hold can press this digit to switch to listening
187 to this music on hold class.
191 * AEL upgraded to use the Gosub with Arguments instead
192 of Macro application, to hopefully reduce the problems
193 seen with the artificially low stack ceiling that
194 Macro bumps into. Macros can only call other Macros
195 to a depth of 7. Tests run using gosub, show depths
196 limited only by virtual memory. A small test demonstrated
197 recursive call depths of 100,000 without problems.
198 -- in addition to this, all apps that allowed a macro
199 to be called, as in Dial, queues, etc, are now allowing
200 a gosub call in similar fashion.
201 * AEL now generates LOCAL(argname) declarations when it
202 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
203 etc. That makes the arguments local in scope. The user
204 can define their own local variables in macros, now,
205 by saying "local myvar=someval;" or using Set() in this
206 fashion: Set(LOCAL(myvar)=someval); ("local" is now
208 * utils/conf2ael introduced. Will convert an extensions.conf
209 file into extensions.ael. Very crude and unfinished, but
210 will be improved as time goes by. Should be useful for a
211 first pass at conversion.
212 * aelparse will now read extensions.conf to see if a referenced
213 macro or context is there before issueing a warning.
215 Zaptel channel driver (chan_zap) Changes
216 ----------------------------------------
217 * SS7 support in chan_zap (via libss7 library)
218 * In India, some carriers transmit CID via dtmf. Some code has been added
219 that will handle some situations. The cidstart=polarity_IN choice has been added for
220 those carriers that transmit CID via dtmf after a polarity change.
221 * CID matching information is now shown when doing 'dialplan show'.
222 * Added zap show version CLI command to chan_zap.
223 * Added setvar support to zapata.conf channel entries.
227 * H323 remote hold notification support added (by NOTIFY message
228 and/or H.450 supplementary service)
230 Call Features (res_features) Changes
231 ------------------------------------
232 * Added the parkedcalltransfers option to features.conf
233 * The built-in method for doing attended transfers has been updated to
234 include some new options that allow you to have the transferee sent
235 back to the person that did the transfer if the transfer is not successful.
236 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
237 in features.conf.sample.
238 * Added support for configuring named groups of custom call features in
239 features.conf. This means that features can be written a single time, and
240 then mapped into groups of features for different key mappings or easier
243 Language Support Changes
244 ------------------------
245 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
246 * Added support for the Hungarian language for saying numbers, dates, and times.
250 * Added the bindaddr option to gtalk.conf.
251 * Argument support for Gosub application
252 * Ability to set process limits without restarting Asterisk
253 * Proper codec support in chan_skinny.
254 * Ability to use libcap to set high ToS bits when non-root
255 on Linux. If configure is unable to find libcap then you
256 can use --with-cap to specify the path.
257 * Added rotatetimestamp option to logger.conf which will use
258 the time to name the logger files instead of sequence number.
259 * Added Masquerade manager event for when a masquerade happens between
261 * From the to-do lists: straighten out the app timeout args:
262 Wait() app now really does 0.3 seconds- was truncating arg to an int.
263 WaitExten() same as Wait().
264 Congestion() - Now takes floating pt. argument.
265 Busy() - now takes floating pt. argument.
266 Read() - timeout now can be floating pt.
267 WaitForRing() now takes floating pt timeout arg.
268 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
269 * Added maxfiles option to options section of asterisk.conf which allows you to specify
270 what Asterisk should set as the maximum number of open files when it loads.
271 * Added the jittertargetextra configuration option.
272 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
273 * Added 's' option to Page application.
274 * Added 'E' and 'V' commands to ExternalIVR.
275 * Added 'o' and 'X' options to Chanspy.
276 * Added a new CDR module, cdr_sqlite3_custom.
277 * The cdr_manager module has a [mappings] feature, like cdr_custom,
278 to add fields to the manager event from the CDR variables.
279 * Added a new realtime configuration module, res_config_sqlite
280 * Added a new dialplan application, Bridge, which allows you to bridge the
281 calling channel to any other active channel on the system.
282 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
283 configuration files for the IP channel drivers. The new option is "cos".
284 This information is also documented in doc/qos.tex, or the IP Quality of Service
285 section of asterisk.pdf.
286 * The device state functionality in the Local channel driver has been updated
287 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
288 to just UNKNOWN if the extension exists.
289 * When originating a call using AMI or pbx_spool that fails the reason for failure
290 will now be available in the failed extension using the REASON dialplan variable.