1 ==============================================================================
3 === This file documents the new and/or enhanced functionality added in
4 === the Asterisk versions listed below. This file does NOT include
5 === changes in behavior that would not be backwards compatible with
6 === previous versions; for that information see the UPGRADE.txt file
7 === and the other UPGRADE files for older releases.
9 ==============================================================================
11 ------------------------------------------------------------------------------
12 --- Functionality changes from Asterisk 1.8 to Asterisk 1.10 -----------------
13 ------------------------------------------------------------------------------
15 Asterisk Manager Interface
16 --------------------------
17 * PeerStatus now includes Address and Port.
20 --------------------------
21 * The HTTP Server can bind to IPv6 addresses.
23 ------------------------------------------------------------------------------
24 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
25 ------------------------------------------------------------------------------
29 * Added preferred_codec_only option in sip.conf. This feature limits the joint
30 codecs sent in response to an INVITE to the single most preferred codec.
31 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
32 to be used for the outgoing call. It must be one of the codecs configured
34 * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
35 to be used for holding a private key. If tlsprivatekey is not specified,
36 tlscertfile is searched for both public and private key.
37 * Added tlsclientmethod option to sip.conf. This allows the protocol for
38 outbound client connections to be specified.
39 * The sendrpid parameter has been expanded to include the options
40 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
41 header to be sent (equivalent to setting sendrpid=yes) and setting
42 sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
43 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
44 is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
45 since the call will fail if Asterisk does not process the incoming SDP, Asterisk
46 will accept the SDP even if the SDP version number is not properly incremented,
47 but will generate a warning in the log indicating that the SIP peer that sent
48 the SDP should have the 'ignoresdpversion' option set.
49 * The 'nat' option has now been been changed to have yes, no, force_rport, and
50 comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
51 symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
52 remote side requests it and disables symmetric RTP support. Setting it to
53 force_rport forces RFC 3581 behavior and disables symmetric RTP support.
54 Setting it to comedia enables RFC 3581 behavior if the remote side requests it
55 and enables symmetric RTP support.
56 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
57 response. This permits the master channel to know how each channel dialled
58 in a multi-channel setup resolved in an individual way.
59 * Added 'externtcpport' and 'externtlsport' options to allow custom port
60 configuration for the externip and externhost options when tcp or tls is used.
61 * Added support for message body (stored in content variable) to SIP NOTIFY message
62 accessible via AMI and CLI.
63 * Added 'media_address' configuration option which can be used to explicitly specify
64 the IP address to use in the SDP for media (audio, video, and text) streams.
65 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
66 that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
68 * Added 'use_q850_reason' configuration option for generating and parsing
69 if available Reason: Q.850;cause=<cause code> header. It is implemented
70 in some gateways for better passing PRI/SS7 cause codes via SIP.
71 * When dialing SIP peers, a new component may be added to the end of the dialstring
72 to indicate that a specific remote IP address or host should be used when dialing
73 the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
74 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
75 ability to selectively force bridged channels to also be encrypted is also
76 implemented. Branching in the dialplan can be done based on whether or not
77 a channel has secure media and/or signaling.
78 * Added directmediapermit/directmediadeny to limit which peers can send direct media
80 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
81 Charge messages to snom phones.
82 * Added support for G.719 media streams.
83 * Added support for 16khz signed linear media streams.
84 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
85 RTP has been outfitted with the same abilities.
86 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
87 available in device configurations as well as in the dial plan.
88 * Addition of the 'subscribe_network_change' option for turning on and off
89 res_stun_monitor module support in chan_sip.
90 * Addition of the 'auth_options_requests' option for turning on and off
91 authentication for OPTIONS requests in chan_sip.
96 * Added rtsavesysname option into iax.conf to allow the systname to be saved
98 * Added the ability for chan_iax2 to inform the dialplan whether or not
99 encryption is being used. This interoperates with the SIP SRTP implementation
100 so that a secure SIP call can be bridged to a secure IAX call when the
101 dialplan requires bridged channels to be "secure".
102 * Addition of the 'subscribe_network_change' option for turning on and off
103 res_stun_monitor module support in chan_iax.
108 * Added ability to preset channel variables on indicated lines with the setvar
109 configuration option. Also, clearvars=all resets the list of variables back
111 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
112 See configs/res_pktccops.conf for more information.
114 XMPP Google Talk/Jingle changes
115 -------------------------------
116 * Added the externip option to gtalk.conf.
117 * Added the stunaddr option to gtalk.conf which allows for the automatic
118 retrieval of the external ip from a stun server.
122 * Added 'p' option to PickupChan() to allow for picking up channel by the first
123 match to a partial channel name.
124 * Added .m3u support for Mp3Player application.
125 * Added progress option to the app_dial D() option. When progress DTMF is
126 present, those values are sent immediately upon receiving a PROGRESS message
127 regardless if the call has been answered or not.
128 * Added functionality to the app_dial F() option to continue with execution
129 at the current location when no parameters are provided.
130 * Added the 'a' option to app_dial to answer the calling channel before any
131 announcements or macros are executed.
132 * Modified app_dial to set answertime when the called channel answers even if
133 the called channel hangs up during playback of an announcement.
134 * Modified app_dial 'r' option to support an additional parameter to play an
135 indication tone from indications.conf
136 * Added c() option to app_chanspy. This option allows custom DTMF to be set
137 to cycle through the next available channel. By default this is still '*'.
138 * Added x() option to app_chanspy. This option allows DTMF to be set to
139 exit the application.
140 * The Voicemail application has been improved to automatically ignore messages
141 that only contain silence.
142 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
143 associated mailbox(es) to be greetings-only.
144 * The ChanSpy application now has the 'S' option, which makes the application
145 automatically exit once it hits a point where no more channels are available
147 * The ChanSpy application also now has the 'E' option, which spies on a single
148 channel and exits when that channel hangs up.
149 * The MeetMe application now turns on the DENOISE() function by default, for
150 each participant. In our tests, this has significantly decreased background
151 noise (especially noisy data centers).
152 * Voicemail now permits storage of secrets in a separate file, located in the
153 spool directory of each individual user. The control for this is located in
154 the "passwordlocation" option in voicemail.conf. Please see the sample
155 configuration for more information.
156 * The ChanIsAvail application now exposes the returned cause code using a separate
157 variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
158 * Added 'd' option to app_followme. This option disables the "Please hold"
160 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
161 received will terminate recording.
162 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
163 Previously the folder could only be set per context, but has now been extended
164 using the imapfolder option.
165 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
166 * Voicemail now allows the pager date format to be specified separately from the
168 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
169 to allow joining, leaving, and sending text to group chats.
170 * MeetMe has a new option 'G' to play an announcement before joining a conference.
171 * Page has a new option 'A(x)' which will playback an announcement simultaneously
172 to all paged phones (and optionally excluding the caller's one using the new
173 option 'n') before the call is bridged.
174 * The 'f' option to Dial has been augmented to take an optional argument. If no
175 argument is provided, the 'f' option works as it always has. If an argument is
176 provided, then the connected party information of all outgoing channels created
177 during the Dial will be set to the argument passed to the 'f' option.
178 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
180 * The OSP lookup application adds in/outbound network ID, optional security,
181 number portability, QoS reporting, destination IP port, custom info and service
183 * Added new application VMSayName that will play the recorded name of the voicemail
184 user if it exists, otherwise will play the mailbox number.
185 * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
186 retrieve state for a particular bridge, where <name> is the conference name
187 * app_directory now allows exiting at any time using the operator or pound key.
188 * Voicemail now supports setting a locale per-mailbox.
189 * Two new applications are provided for declining counting phrases in multiple
190 languages. See the application notes for SayCountedNoun and SayCountedAdj for
192 * Voicemail now runs the externnotify script when pollmailboxes is activated and
194 * Voicemail now includes rdnis within msgXXXX.txt file.
195 * Added 'D' command to ExternalIVR full details in doc/externalivr.txt
199 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
200 over SRV records associated with a specific service. From the CLI, type
201 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
202 details on how these may be used.
203 * PITCH_SHIFT dialplan function added. This function can be used to modify the
204 pitch of a channel's tx and rx audio streams.
205 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
206 setting various connected line and redirecting party information.
207 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
208 support ISDN subaddressing.
209 * The CHANNEL() function now supports the "name" and "checkhangup" options.
210 * For DAHDI channels, the CHANNEL() dialplan function now allows
211 the dialplan to request changes in the configuration of the active
212 echo canceller on the channel (if any), for the current call only.
215 exten => s,n,Set(CHANNEL(echocan_mode)=off)
217 The possible values are:
219 on - normal mode (the echo canceller is actually reinitialized)
221 fax - FAX/data mode (NLP disabled if possible, otherwise completely
223 voice - voice mode (returns from FAX mode, reverting the changes that
224 were made when FAX mode was requested)
225 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
226 and setting variables on the channel which created the current channel.
227 Administrators should take care to avoid naming conflicts, when multiple
228 channels are dialled at once, especially when used with the Local channel
229 construct (which all could set variables on the master channel). Usage
230 of the HASH() dialplan function, with the key set to the name of the slave
231 channel, is one approach that will avoid conflicts.
232 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
234 * func_odbc now allows multiple row results to be retrieved without using
235 mode=multirow. If rowlimit is set, then additional rows may be retrieved
236 from the same query by using the name of the function which retrieved the
237 first row as an argument to ODBC_FETCH().
238 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
239 dialplan. This function returns the content of the received message.
240 * Added REPLACE, which searches a given variable name for a set of characters,
241 then either replaces them with a single character or deletes them.
242 * Added PASSTHRU, which literally passes the same argument back as its return
243 value. The intent is to be able to use a literal string argument to
244 functions that currently require a variable name as an argument.
245 * HASH-associated variables now can be inherited across channel creation, by
246 prefixing the name of the hash at assignment with the appropriate number of
247 underscores, just like variables.
248 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
249 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
250 whether or not channels that are bridged to the current channel will be
251 required to have secure signaling and/or media.
252 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
253 the current channel has secure signaling and/or media.
254 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
255 "no_media_path" option.
256 Returns "0" if there is a B channel associated with the call.
257 Returns "1" if no B channel is associated with the call. The call is either
258 on hold or is a call waiting call.
259 * Added option to dialplan function CDR(), the 'f' option
260 allows for high resolution times for billsec and duration fields.
261 * FILE() now supports line-mode and writing.
262 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
263 * FRAME_TRACE(), for tracking internal ast_frames on a channel.
267 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
268 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
269 and is set when a dynamic feature is triggered.
270 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
271 to dynamically create a new parking lot matching the value this varible is
273 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
274 features.conf that should be the base for dynamic parkinglots.
275 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
276 parkinglot should have.
277 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
282 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
284 * Added 'R' option to app_queue. This option stops moh and indicates ringing
285 to the caller when an Agent's phone is ringing. This can be used to indicate
286 to the caller that their call is about to be picked up, which is nice when
287 one has been on hold for an extened period of time.
288 * A new config option, penaltymemberslimit, has been added to queues.conf.
289 When set this option will disregard penalty settings when a queue has too
291 * A new option, 'I' has been added to both app_queue and app_dial.
292 By setting this option, Asterisk will not update the caller with
293 connected line changes or redirecting party changes when they occur.
294 * A 'relative-peroidic-announce' option has been added to queues.conf. When
295 enabled, this option will cause periodic announce times to be calculated
296 from the end of announcements rather than from the beginning.
297 * The autopause option in queues.conf can be passed a new value, "all." The
298 result is that if a member becomes auto-paused, he will be paused in all
299 queues for which he is a member, not just the queue that failed to reach
301 * Added dialplan function QUEUE_EXISTS to check if a queue exists
302 * The queue logger now allows events to optionally propagate to a file,
303 even when realtime logging is turned on. Additionally, realtime logging
304 supports sending the event arguments to 5 individual fields, although it
305 will fallback to the previous data definition, if the new table layout is
308 mISDN channel driver (chan_misdn) changes
309 ----------------------------------------
310 * Added display_connected parameter to misdn.conf to put a display string
311 in the CONNECT message containing the connected name and/or number if
312 the presentation setting permits it.
313 * Added display_setup parameter to misdn.conf to put a display string
314 in the SETUP message containing the caller name and/or number if the
315 presentation setting permits it.
316 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
317 indicate the dialplan settings are to be obtained from the asterisk
319 * Made misdn.conf parameter callerid accept the "name" <number> format
320 used by the rest of the system.
321 * Made use the nationalprefix and internationalprefix misdn.conf
322 parameters to prefix any received number from the ISDN link if that
323 number has the corresponding Type-Of-Number. NOTE: This includes
324 comparing the incoming call's dialed number against the MSN list.
325 * Added the following new parameters: unknownprefix, netspecificprefix,
326 subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
327 received number from the ISDN link if that number has the corresponding
329 * Added new dialplan application misdn_command which permits controlling
330 the CCBS/CCNR functionality.
331 * Added new dialplan function mISDN_CC which permits retrieval of various
332 values from an active call completion record.
333 * For PTP, you should manually send the COLR of the redirected-to party
334 for an incomming redirected call if the incoming call could experience
335 further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
336 set the REDIRECTING(to-pres) to the COLR. A call has been redirected
337 if the REDIRECTING(from-num) is not empty.
338 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
339 option on all of the REDIRECTING statements before dialing the
340 redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
341 and the REDIRECTING(from-xxx,i) values. The PTP call will update the
342 redirecting-to presentation (COLR) when it becomes available.
343 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
346 thirdparty mISDN enhancements
347 -----------------------------
348 mISDN has been modified by Digium, Inc. to greatly expand facility message
350 * Enhanced COLP support for call diversion and transfer.
353 The latest modified mISDN v1.1.x based version is available at:
354 http://svn.digium.com/svn/thirdparty/mISDN/trunk
355 http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
357 Tagged versions of the modified mISDN code are available under:
358 http://svn.digium.com/svn/thirdparty/mISDN/tags
359 http://svn.digium.com/svn/thirdparty/mISDNuser/tags
361 libpri channel driver (chan_dahdi) DAHDI changes
362 -------------------------------------------
363 * The channel variable PRIREDIRECTREASON is now just a status variable
364 and it is also deprecated. Use the REDIRECTING(reason) dialplan function
365 to read and alter the reason.
366 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
367 redirected-to party for an incomming redirected call if the incoming call
368 could experience further redirects. Just set the
369 REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
370 to the COLR. A call has been redirected if the REDIRECTING(count) is not
372 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
373 use the inhibit(i) option on all of the REDIRECTING statements before
374 dialing the redirected-to party. You still have to set the
375 REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
376 will update the redirecting-to presentation (COLR) when it becomes available.
377 * Added the ability to ignore calls that are not in a Multiple Subscriber
378 Number (MSN) list for PTMP CPE interfaces.
379 * Added dynamic range compression support for dahdi channels. It is
380 configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
381 * Added support for ISDN calling and called subaddress with partial support
382 for connected line subaddress.
383 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
384 * Added handling of received HOLD/RETRIEVE messages and the optional ability
385 to transfer a held call on disconnect similar to an analog phone.
386 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
387 Will reroute/deflect an outgoing call when receive the message.
388 Can use the DAHDISendCallreroutingFacility to send the message for the
390 * Added standard location to add options to chan_dahdi dialing:
391 Dial(DAHDI/g1[/extension[/options]])
394 R Reverse charging indication
395 * Added Reverse Charging Indication (Collect calls) send/receive option.
396 Send reverse charging in SETUP message with the chan_dahdi R dialing option.
397 Dial(DAHDI/g1/extension/R)
398 Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
399 (requires latest LibPRI)
400 * Added ability to send/receive keypad digits in the SETUP message.
401 Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
402 dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
403 Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
404 (requires latest LibPRI)
405 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
406 to eliminate tromboned calls. A tromboned call goes out an interface and comes
407 back into the same interface. Tromboned calls happen because of call routing,
408 call deflection, call forwarding, and call transfer.
409 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
410 * Added the ability to support call waiting calls. (The SETUP has no B channel
412 * Added Malicious Call ID (MCID) event to the AMI call event class.
413 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
415 Asterisk Manager Interface
416 --------------------------
417 * The Hangup action now accepts a Cause header which may be used to
418 set the channel's hangup cause.
419 * sslprivatekey option added to manager.conf and http.conf. Adds the ability
420 to specify a separate .pem file to hold a private key. By default sslcert
421 is used to hold both the public and private key.
422 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
423 for options containing the 'tls' prefix. For example, 'sslenable' is now
424 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
425 across all .conf files. All affected sample.conf files have been modified to
426 reflect this change. Previous options such as 'sslenable' still work,
427 but options with the 'tls' prefix are preferred.
428 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
429 in a channel. (res_mutestream.so)
430 * The configuration file manager.conf now supports a channelvars option, which
431 specifies a list of channel variables to include in each channel-oriented
433 * The redirect command now has new parameters ExtraContext, ExtraExtension,
434 and ExtraPriority to allow redirecting the second channel to a different
435 location than the first.
436 * Added new event "JabberStatus" in the Jabber module to monitor buddies
438 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
439 in a MixMonitor recording.
440 * The 'iax2 show peers' output is now similar to the expected output of
442 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
444 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
445 AOC-E messages on a channel.
446 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
447 conform more closely to similar events.
448 * Added a new eventfilter option per user to allow whitelisting and blacklisting
450 * Added optional parkinglot variable for park command.
452 Channel Event Logging
453 ---------------------
454 * A new interface, CEL, is introduced here. CEL logs single events, much like
455 the AMI, but it differs from the AMI in that it logs to db backends much
456 like CDR does; is based on the event subsystem introduced by Russell, and
457 can share in all its benefits; allows multiple backends to operate like CDR;
458 is specialized to event data that would be of concern to billing sytems,
459 like CDR. Backends for logging and accounting calls have been produced,
460 but a new CDR backend is still in development.
464 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
465 linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
466 etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
467 * Multiple files and formats can now be specified in cdr_custom.conf.
468 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
469 See configs/cdr_syslog.conf.sample for more information.
470 * A 'sequence' field has been added to CDRs which can be combined with
471 linkedid or uniqueid to uniquely identify a CDR.
472 * Handling of billsec and duration field has changed. If your table definition
473 specifies those fields as float,double or similar they will now be logged with
474 microsecond accuracy instead of a whole integer.
476 Calendaring for Asterisk
477 ------------------------
478 * A new set of modules were added supporing calendar integration with Asterisk.
479 Dialplan functions for reading from and writing to calendars are included,
480 as well as the ability to execute dialplan logic upon calendar event notifications.
481 iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
482 Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
483 Exchange Server 2007+ with full write and attendee support) are supported (Exchange
484 2003 support does not support forms-based authentication).
486 Call Completion Supplementary Services for Asterisk
487 ---------------------------------------------------
488 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
489 DAHDI/ISDN supports call completion for the following switch types:
490 EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
491 See doc/CCSS_architecture.pdf and doc/tex/ccss.tex(asterisk.pdf) for details.
493 Multicast RTP Support
494 ---------------------
495 * A new RTP engine and channel driver have been added which supports Multicast RTP.
496 The channel driver can be used with the Page application to perform multicast RTP
497 paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
498 Type can be either basic or linksys.
499 Destination is the IP address and port for the RTP packets.
500 Control address is specific to the linksys type and is used for sending the control
501 packets unique to them.
503 Security Events Framework
504 -------------------------
505 * Asterisk has a new C API for reporting security events. The module res_security_log
506 sends these events to the "security" logger level. Currently, AMI is the only
507 Asterisk component that reports security events. However, SIP support will be
508 coming soon. For more information on the security events framework, see the
509 "Security Events" chapter of the included documentation - doc/tex/asterisk.pdf.
513 * A technology independent fax frontend (res_fax) has been added to Asterisk.
514 * A spandsp based fax backend (res_fax_spandsp) has been added.
515 * The app_fax module has been deprecated in favor of the res_fax module and
516 the new res_fax_spandsp backend.
517 * The SendFAX and ReceiveFAX applications now send their log messages to a
518 'fax' logger level, instead of to the generic logger levels. To see these
519 messages, the system's logger.conf file will need to direct the 'fax' logger
520 level to one or more destinations; the logger.conf.sample file includes an
521 example of how to do this. Note that if the 'fax' logger level is *not*
522 directed to at least one destination, log messages generated by these
523 applications will be lost, and that if the 'fax' logger level is directed to
524 the console, the 'core set verbose' and 'core set debug' CLI commands will
525 have no effect on whether the messages appear on the console or not.
529 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
530 Now, in order to enable transmitting silence during record the transmit_silence
531 option should be used. transmit_silence_during_record remains a valid option, but
532 defaults to the behavior of the transmit_silence option.
533 * Addition of the Unit Test Framework API for managing registration and execution
534 of unit tests with the purpose of verifying the operation of C functions.
535 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
536 XMPP text messages to the remote JID.
537 * Modules.conf has a new option - "require" - that marks a module as critical for
538 the execution of Asterisk.
539 If one of the required modules fail to load, Asterisk will exit with a return
541 * An 'X' option has been added to the asterisk application which enables #exec support.
542 This allows #exec to be used in asterisk.conf.
543 * jabber.conf supports a new option auth_policy that toggles auto user registration.
544 * A new lockconfdir option has been added to asterisk.conf to protect the
545 configuration directory (/etc/asterisk by default) during reloads.
546 * The parkeddynamic option has been added to features.conf to enable the creation
547 of dynamic parkinglots.
548 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
549 the reportalarms config option.
550 * chan_dahdi supports dialing configuring and dialing by device file name.
551 DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
552 it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
553 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
554 False by default. If set, chan_dahdi will ignore failed 'channel' entries.
555 Handy for the above name-based syntax as it does not depend on
556 initialization order.
557 * The Realtime dialplan switch now caches entries for 1 second. This provides a
558 significant increase in performance (about 3X) for installations using this switchtype.
559 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
560 AIS. For more information, please see doc/distributed_devstate-XMPP.txt
561 * The addition of G.719 pass-through support.
562 * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
563 during device configuration.
564 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
565 have less than 3 lines on the LCD.
566 * Realtime now supports database failover. See the sample extconfig.conf for details.
567 * The addition of improved translation path building for wideband codecs. Sample
568 rate changes during translation are now avoided unless absolutely necessary.
569 * The addition of the res_stun_monitor module for monitoring and reacting to network
570 changes while behind a NAT.
574 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
575 optionally accept a filename, to apply the setting only to the code generated from
576 that source file when Asterisk was built. However, there are some modules in Asterisk
577 that are composed of multiple source files, so this did not result in the behavior
578 that users expected. In this version, 'core set debug' and 'core set verbose'
579 can optionally accept *module* names instead (with or without the .so extension),
580 which applies the setting to the entire module specified, regardless of which source
581 files it was built from.
582 * New 'manager show settings' command showing the current settings loaded from
584 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
585 the channel hangup request to all channels.
586 * Added a "core reload" CLI command that executes a global reload of Asterisk.
588 ------------------------------------------------------------------------------
589 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
590 ------------------------------------------------------------------------------
594 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
595 Snom phones use this for call pickup of extensions that the phone is
597 * Added support for setting the domain in the URI for caller of an
598 outbound call by using the SIPFROMDOMAIN channel variable.
599 * Added a new configuration option "remotesecret" for authentication to
600 remote services. For backwards compatibility, "secret" still has the
601 same function as before, but now you can configure both a remote secret and a
602 local secret for mutual authentication.
603 * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
604 the sound will be played to the target of an attended transfer
605 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
606 finer control over how many peers Asterisk will qualify and the gap between them
607 when all peers need to be qualified at the same time.
608 * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
609 (either globally or for a specific peer), chan_sip will treat any SDP data
610 it receives as new data and update the media stream accordingly. By
611 default, Asterisk will only modify the media stream if the SDP session
612 version received is different from the current SDP session version. This
613 option is required to interoperate with devices that have non-standard SDP
614 session version implementations (observed with Microsoft OCS). This option
615 is disabled by default.
616 * The parsing of register => lines in sip.conf has been modified to allow a port
617 to be present in the "user" portion. Please see the sip.conf.sample file for more
619 * Added support for subscribing to MWI on a remote server and making the status available
620 as a mailbox. Please see the sip.conf.sample file for more information.
621 * Added a function to remove SIP headers added in the dialplan before the
622 first INVITE is generated - SIPRemoveHeader()
623 * Channel variables set with setvar= in a device configuration is now
624 set both for inbound and outbound calls.
625 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
629 * Added immediate option to iax.conf
630 * Added forceencryption option to iax.conf
631 * Added Encryption and Trunk status to manager command "iaxpeers"
635 * The configuration file now holds separate sections for devices and lines.
636 Please have a look at configs/skinny.conf.sample and change your skinny.conf
641 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
642 support for LibOpenR2. http://www.libopenr2.org/
643 * The UK option waitfordialtone has been added for use with BT analog
645 * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
646 is used in conjunction with the 'faxdetect' configuration option. When
647 'faxbuffers' is used and fax tones are detected, the channel will dynamically
648 switch to the configured faxbuffers policy. For example, to use 6 buffers
649 and a 'full' buffer policy for a fax transmission, add:
651 The faxbuffers configuration will be in affect until the call is torn down.
652 * Added service message support for 4ESS/5ESS switches.
656 * For DAHDI channels, the CHANNEL() dialplan function now
657 supports changing the channel's buffer policy (for the current
658 call only), using this syntax:
660 exten => s,n,Set(CHANNEL(buffers)=6,full)
662 This would change the channel to the 'full' buffer policy and
663 6 (six) buffers. Possible options for this setting are the same
664 as those in chan_dahdi.conf.
665 * Added a new dialplan function, CURLOPT, which permits setting various
666 options that may be useful with the CURL dialplan function, such as
667 cookies, proxies, connection timeouts, passwords, etc.
668 * Permit the syntax and synopsis fields of the corresponding dialplan
669 functions to be individually set from func_odbc.conf.
670 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
671 * func_odbc now may specify an insert query to execute, when the write query
672 affects 0 rows (usually indicating that no such row exists).
673 * Added a new dialplan function, LISTFILTER, which permits removing elements
674 from a set list, by name. Uses the same general syntax as the existing CUT
675 and FIELDQTY dialplan functions, which also manage lists.
676 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
677 obtaining realtime data from the dialplan.
678 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
679 a subroutine when using the GoSub() and Return() applications.
680 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
681 of "core show function AUDIOHOOK_INHERIT" from the CLI
682 * Added AES_ENCRYPT. For information on its use, please see the output
683 of "core show function AES_ENCRYPT" from the CLI
684 * Added AES_DECRYPT. For information on its use, please see the output
685 of "core show function AES_DECRYPT" from the CLI
686 * func_odbc now supports database transactions across multiple queries.
690 * Scheduled meetme conferences may now have their end times extended by
692 * app_authenticate now gives the ability to select a prompt other than
694 * app_directory now pays attention to the searchcontexts setting in
695 voicemail.conf and will look through all contexts, if no context is
696 specified in the initial argument.
697 * A new application, Originate, has been introduced, that allows asynchronous
698 call origination from the dialplan.
699 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
700 in addition to the setting in the "general" context.
701 * Added ConfBridge dialplan application which does conference bridges without
702 DAHDI. For information on its use, please see the output of
703 "core show application ConfBridge" from the CLI.
707 * The Asterisk CLI has a new command, "channel redirect", which is similar in
708 operation to the AMI Redirect action.
709 * extensions.conf now allows you to use keyword "same" to define an extension
710 without actually specifying an extension. It uses exactly the same pattern
711 as previously used on the last "exten" line. For example:
712 exten => 123,1,NoOp(something)
713 same => n,SomethingElse()
714 * musiconhold.conf classes of type 'files' can now use relative directory paths,
715 which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
716 * All deprecated CLI commands are removed from the sourcecode. They are now handled
717 by the new clialiases module. See cli_aliases.conf.sample file.
718 * Times within timespecs are now accurate down to the minute. This is a change
719 from historical Asterisk, which only provided timespecs rounded to the nearest
720 even (read: evenly divisible by 2) minute mark.
721 * The realtime switch now supports an option flag, 'p', which disables searches for
723 * In addition to a time range and date range, timespecs now accept a 5th optional
724 argument, timezone. This allows you to perform time checks on alternate
725 timezones, especially if those daylight savings time ranges vary from your
726 machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
728 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
729 give you the correct output for an asterisk box behind nat. It will give you the
730 externhost and localnet settings.
731 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
732 can connect calls in passthrough mode, as well as record and play back files.
733 * Successful and unsuccessful call pickup can now be alerted through sounds, by
734 using pickupsound and pickupfailsound in features.conf.
735 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
736 This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
737 instead of the /var/run/asterisk.pid where it used to be. This will make
738 installs as non-root easier to manage.
743 * The cdr.conf file must exist and be correctly programmed in order for CDR records to
744 be written; they will no longer be explicitly written.
746 Asterisk Manager Interface
747 --------------------------
748 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
749 a non-empty value) in your request. If you do this, any pending AMI events will
750 *not* be included in the response to your request as they would normally, but
751 will be left in the event queue for the next request you make to retrieve. For
752 some applications, this will allow you to guarantee that you will only see
753 events in responses to 'WaitEvent' actions, and can better know when to expect them.
754 To know whether the Asterisk server supports this header or not, your client can
755 inspect the first response back from the server to see if it includes this header:
757 Pragma: SuppressEvents
759 If this is included, the server supports event suppression.
761 * Added 4 new Actions to list skinny device(s) and line(s)
767 LDAP Schema File Additions
768 --------------------------
769 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
770 to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
772 - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
773 - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
774 - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
775 * Removed redundant IPaddr (there's already IPAddress)
776 - Gives more configuration Flags for SIP-Users available (tested)
777 - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
778 without extensibleObject (which really should be the last resort); gives
779 also additional possibilities for LDAP-filter
781 ------------------------------------------------------------------------------
782 --- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
783 ------------------------------------------------------------------------------
785 Device State Handling
786 ---------------------
787 * The event infrastructure in Asterisk got another big update to help support
788 distributed events. It currently supports distributed device state and
789 distributed Voicemail MWI (Message Waiting Indication). A new module has
790 been merged, res_ais, which facilitates communicating events between servers.
791 It uses the SAForum AIS (Service Availability Forum Application Interface
792 Specification) CLM (Cluster Management) and EVT (Event) services to maintain
793 a cluster of Asterisk servers, and to share events between them. For more
794 information on setting this up, see doc/distributed_devstate.txt.
798 * Added a new dialplan function, AST_CONFIG(), which allows you to access
799 variables from an Asterisk configuration file.
800 * The JACK_HOOK function now has a c() option to supply a custom client name.
801 * Added two new dialplan functions from libspeex for audio gain control and
802 denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
803 rx directions of a channel from the dialplan.
804 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
805 based on other parameters. The default is still to search based on the
806 forwarding station ID. However, there are new options that allow you to search
807 based on the message desk terminal ID, or the message desk number.
808 * TIMEOUT() has been modified to be accurate down to the millisecond.
809 * ENUM*() functions now include the following new options:
810 - 'u' returns the full URI and does not strip off the URI-scheme.
811 - 's' triggers ISN specific rewriting
812 - 'i' looks for branches into an Infrastructure ENUM tree
813 - 'd' for a direct DNS lookup without any flipping of digits.
814 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
815 * CHANNEL() now has options for the maximum, minimum, and standard or normal
816 deviation of jitter, rtt, and loss for a call using chan_sip.
818 DAHDI channel driver (chan_dahdi) Changes
819 ----------------------------------------
820 * Channels can now be configured using named sections in chan_dahdi.conf, just
821 like other channel drivers, including the use of templates.
822 * The default for pridialplan has changed from 'national' to 'unknown'.
826 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
827 to something that matches the pattern a hint will be created using the contents
828 and variables evaluated.
829 * Dialplan matching has been extended to allow an extension to return to the
830 PBX core to wait for more digits. This is done by using the new dialplan
831 application called "Incomplete". This will permit a whole new level of
832 extension control, by giving the administrator more control over early
833 matches employing one of the short-circuit pattern match operators. Note
834 that custom applications can trigger this same behavior by returning the
835 special value AST_PBX_INCOMPLETE.
839 * Directory now permits both first and last names to be matched at the same
840 time. In addition, the number of digits to enter of the name can be set in
841 the arguments to Directory; previously, you could enter only 3, regardless
842 of how many names are in your company. For large companies, this should be
844 * Voicemail now permits a mailbox setting to wrap around from first to last
845 messages, if the "messagewrap" option is set to a true value.
846 * Voicemail now permits an external script to be run, for password validation.
847 The script should output "VALID" or "INVALID" on stdout, depending upon the
848 wish to validate or invalidate the password given. Arguments are:
849 "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
851 * Dial has a new option: F(context^extension^pri), which permits a callee to
852 continue in the dialplan, at the specified label, if the caller hangs up.
853 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
854 technology name (e.g. SIP, IAX, etc) of the channel being spied on.
855 * The Jack application now has a c() option to supply a custom client name.
856 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
857 like the pre-existing whisper mode, except that the spy can also talk to the
858 participant on the bridged channel as well.
859 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
860 to be spoken instead of the channel name or number. For more information on the
861 use of this option, issue the command "core show application ChanSpy" from the
863 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
864 spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
865 words, if using the 'd' option, it is not possible to enter a number to append to
866 the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
867 change to whisper mode, and pressing 6 will change to barge mode.
868 * ExternalIVR now takes several options that affect the way it performs, as
869 well as having several new commands. Please see doc/externalivr.txt for the
870 complete documentation.
871 * Added ability to communicate over a TCP socket instead of forking a child process for the
872 ExternalIVR application.
873 * ChanIsAvail has a new option, 'a', which will return all available channels instead
874 of just the first one if you give the function more then one channel to check.
875 * PrivacyManager now takes an option where you can specify a context where the
876 given number will be matched. This way you have more control over who is allowed
877 and it stops the people who blindly enter 10 digits.
878 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
879 answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
880 from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
881 original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
882 the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
883 obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
884 * The Dial() application no longer copies the language used by the caller to the callee's
885 channel. If you desire for the caller's channel's language to be used for file playback
886 to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
887 * SendImage() no longer hangs up the channel on error; instead, it sets the
888 status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
889 'UNSUPPORTED'. This change makes SendImage() more consistent with other
891 * Park has a new option, 's', which silences the announcement of the parking space number.
892 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
893 invalid input and will be assumed to mean that no timeout is desired.
897 * Added DNS manager support to registrations for peers referencing peer entries.
898 DNS manager runs in the background which allows DNS lookups to be run asynchronously
899 as well as periodically updating the IP address. These properties allow for
900 better performance as well as recovery in the event of an IP change.
901 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
902 load/reload of large numbers of peers/users by ~40x (for large lists of peers).
903 These changes also provide performance improvements for call setup and tear down.
904 * Added ability to specify registration expiry time on a per registration basis in
906 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
908 * Added t38pt_usertpsource option. See sip.conf.sample for details.
909 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
910 * 'sip show peers' and 'sip show users' display their entries sorted in
911 alphabetical order, as opposed to the order they were in, in the config
913 * Videosupport now supports an additional option, "always", which always sets
914 up video RTP ports, even on clients that don't support it. This helps with
915 callfiles and certain transfers to ensure that if two video phones are
916 connected, they will always share video feeds.
920 * Existing DNS manager lookups extended to check for SRV records.
921 * IAX2 encryption support has been improved to support periodic key rotation
922 within a call for enhanced security. The option "keyrotate" has been
923 provided to disable this functionality to preserve backwards compatibility
924 with older versions of IAX2 that do not support key rotation.
928 * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
929 data tree based on the given <path>.
930 * New CLI command "data show providers" that will display all the registered
932 * New CLI command, "config reload <file.conf>" which reloads any module that
933 references that particular configuration file. Also added "config list"
934 which shows which configuration files are in use.
935 * New CLI commands, "pri show version" and "ss7 show version" that will
936 display which version of libpri and libss7 are being used, respectively.
937 A new API call was added so trunk will now have to be compiled against
938 a versions of libpri and libss7 that have them or it will not know that
939 these libraries exist.
940 * The commands "core show globals", "core set global" and "core set chanvar" has
941 been deprecated in favor of the more semanticly correct "dialplan show globals",
942 "dialplan set chanvar" and "dialplan set global".
943 * New CLI command "dialplan show chanvar" to list all variables associated
944 with a given channel.
948 * Addresses managed by DNS manager now can check to see if there is a DNS
949 SRV record for a given domain and will use that hostname/port if present.
951 AMI - The manager (TCP/TLS/HTTP)
952 --------------------------------
953 * The Status command now takes an optional list of variables to display
954 along with channel status.
955 * The QueueEntry event now also includes the channel's uniqueid
959 * res_odbc no longer has a limit of 1023 total possible unshared connections,
960 as some people were running into this limit. This limit has been increased
965 * The TRANSFER queue log entry now includes the the caller's original
966 position in the transferred-from queue.
967 * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
968 "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
969 as well as an explanation about timeout options in general
970 * Added a new option - C - for forcing the "answered elsewhere" flag on
971 cancellation of calls in to members of the queue. This is to avoid the
972 call to a member of a queue having the call listed as a "missed call".
976 * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
977 adaptive capabilities. What this means in practical terms is that if your
978 realtime table lacks critical fields, Asterisk will now emit warnings to
979 that effect. Also, some of the realtime drivers have the ability (if
980 configured) to automatically add those columns to the table with the
981 correct type and length.
985 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
986 the 'setvar' option to cause a given audio file to be played upon completion
987 of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
988 Skinny channels only.
989 * You can now compile Asterisk against the Hoard Memory Allocator, see doc/hoard.txt
990 for more information.
991 * Config file variables may now be appended to, by using the '+=' append
992 operator. This is most helpful when working with long SQL queries in
993 func_odbc.conf, as the queries no longer need to be specified on a single
995 * CDR config file, cdr.conf, has an added option, "initiatedseconds",
996 which will add a second to the billsec when the ending
997 time is set, if the number in the microseconds field of the end time is
998 greater than the number of microseconds in the answer time. This allows
999 users to count the 'initiated' seconds in their billing records.
1001 ------------------------------------------------------------------------------
1002 --- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
1003 ------------------------------------------------------------------------------
1005 AMI - The manager (TCP/TLS/HTTP)
1006 --------------------------------
1007 * Manager has undergone a lot of changes, all of them documented
1008 in doc/manager_1_1.txt
1009 * Manager version has changed to 1.1
1010 * Added a new action 'CoreShowChannels' to list currently defined channels
1011 and some information about them.
1012 * Added a new action 'SIPshowregistry' to list SIP registrations.
1013 * Added TLS support for the manager interface and HTTP server
1014 * Added the URI redirect option for the built-in HTTP server
1015 * The output of CallerID in Manager events is now more consistent.
1016 CallerIDNum is used for number and CallerIDName for name.
1017 * Enable https support for builtin web server.
1018 See configs/http.conf.sample for details.
1019 * Added a new action, GetConfigJSON, which can return the contents of an
1020 Asterisk configuration file in JSON format. This is intended to help
1021 improve the performance of AJAX applications using the manager interface
1023 * SIP and IAX manager events now use "ChannelType" in all cases where we
1024 indicate channel driver. Previously, we used a mixture of "Channel"
1025 and "ChannelDriver" headers.
1026 * Added a "Bridge" action which allows you to bridge any two channels that
1027 are currently active on the system.
1028 * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
1029 the voicemail users setup.
1030 * Added 'DBDel' and 'DBDelTree' manager commands.
1031 * cdr_manager now reports events via the "cdr" level, separating it from
1032 the very verbose "call" level.
1033 * Manager users are now stored in memory. If you change the manager account
1034 list (delete or add accounts) you need to reload manager.
1035 * Added Masquerade manager event for when a masquerade happens between
1037 * Added "manager reload" command for the CLI
1038 * Lots of commands that only provided information are now allowed under the
1039 Reporting privilege, instead of only under Call or System.
1040 * The IAX* commands now require either System or Reporting privilege, to
1041 mirror the privileges of the SIP* commands.
1042 * Added ability to retrieve list of categories in a config file.
1043 * Added ability to retrieve the content of a particular category.
1044 * Added ability to empty a context.
1045 * Created new action to create a new file.
1046 * Updated delete action to allow deletion by line number with respect to category.
1047 * Added new action insert to add new variable to category at specified line.
1048 * Updated action newcat to allow new category to be inserted in file above another
1050 * Added new event "JitterBufStats" in the IAX2 channel
1051 * Originate now requires the Originate privilege and, if you want to call out
1052 to a subshell, it requires the System privilege, as well. This was done to
1053 enhance manager security.
1054 * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
1055 * New command: Atxfer. See doc/manager_1_1.txt for more details or
1056 manager show command Atxfer from the CLI
1057 * New command: IAXregistry. See doc/manager_1_1.txt for more details or
1058 manager show command IAXregistry from the CLI
1062 * Added the DEVICE_STATE() dialplan function which allows retrieving any device
1063 state in the dialplan, as well as creating custom device states that are
1064 controllable from the dialplan.
1065 * Extend CALLERID() function with "pres" and "ton" parameters to
1066 fetch string representation of calling number presentation indicator
1067 and numeric representation of type of calling number value.
1068 * MailboxExists converted to dialplan function
1069 * A new option to Dial() for telling IP phones not to count the call
1070 as "missed" when dial times out and cancels.
1071 * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
1072 mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
1073 held for any given channel. Also, locks are automatically freed when a
1075 * Added HINT() dialplan function that allows retrieving hint information.
1076 Hints are mappings between extensions and devices for the sake of
1077 determining the state of an extension. This function can retrieve the list
1078 of devices or the name associated with a hint.
1079 * Added EXTENSION_STATE() dialplan function which allows retrieving the state
1081 * Added SYSINFO() dialplan function which allows retrieval of system information
1082 * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
1083 the existence of a dialplan target.
1084 * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
1085 upper and lower case, respectively.
1086 * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
1087 ID for the call (not the Asterisk call ID or unique ID), provided that the
1088 channel driver supports this. For SIP, you get the SIP call-ID for the
1089 bridged channel which you can store in the CDR with a custom field.
1093 * Added CLI permissions, config file: cli_permissions.conf
1094 default is to allow all commands for every local user/group.
1095 Also this new feature added three new CLI commands:
1096 - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
1097 - cli reload permissions
1098 - cli show permissions
1099 * New CLI command "core show hint" (usage: core show hint <exten>)
1100 * New CLI command "core show settings"
1101 * Added 'core show channels count' CLI command.
1102 * Added the ability to set the core debug and verbose values on a per-file basis.
1103 * Added 'queue pause member' and 'queue unpause member' CLI commands
1104 * Ability to set process limits ("ulimit") without restarting Asterisk
1105 * Enhanced "agi debug" to print the channel name as a prefix to the debug
1106 output to make debugging on busy systems much easier.
1107 * New CLI commands "dialplan set extenpatternmatching true/false"
1108 * New CLI command: "core set chanvar" to set a channel variable from the CLI.
1109 * Added an easy way to execute Asterisk CLI commands at startup. Any commands
1110 listed in the startup_commands section of cli.conf will get executed.
1111 * Added a CLI command, "devstate change", which allows you to set custom device
1112 states from the func_devstate module that provides the DEVICE_STATE() function
1113 and handling of the "Custom:" devices.
1114 * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
1115 sorted into the different possible callbacks, with the number of entries
1116 currently scheduled for each. Gives you a feel for how busy the sip channel
1118 * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
1119 * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
1120 (Done by lmadsen, junky and mvanbaak during the devcon 2008)
1124 * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
1125 option is enabled, Asterisk will watch for a CNG tone in the incoming audio
1126 for a received call. If it is detected, the channel will jump to the
1127 'fax' extension in the dialplan.
1128 * The default SIP useragent= identifier now includes the Asterisk version
1129 * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
1130 If set, and the incoming request carries authentication info,
1131 the username to match in the users list is taken from the Digest header
1132 rather than from the From: field. This feature is considered experimental.
1133 * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
1134 since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
1135 * The "localmask" setting was removed in version 1.2 and the reminder about it
1136 being removed is now also removed.
1137 * A new option "busylevel" for setting a level of calls where asterisk reports
1138 a device as busy, to separate it from call-limit. This value is also added
1139 to the SIP_PEER dialplan function.
1140 * A new realtime family called "sipregs" is now supported to store SIP registration
1141 data. If this family is defined, "sippeers" will be used for configuration and
1142 "sipregs" for registrations. If it's not defined, "sippeers" will be used for
1143 registration data, as before.
1144 * The SIPPEER function have new options for port address, call and pickup groups
1145 * Added support for T.140 realtime text in SIP/RTP
1146 * The "checkmwi" option has been removed from sip.conf, as it is no longer
1147 required due to the restructuring of how MWI is handled. See the descriptions
1148 in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
1149 for more information.
1150 * Added rtpdest option to CHANNEL() dialplan function.
1151 * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
1152 * SIP now adds a header to the CANCEL if the call was answered by another phone
1153 in the same dial command, or if the new c option in dial() is used.
1154 * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
1155 states it is not needed. For phones, however, that do require it the "registertrying" option
1156 has been added so it can be enabled.
1157 * A new option called "callcounter" (global/peer/user level) enables call counters needed
1158 for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
1159 used to enable this functionality).
1160 * New settings for timer T1 and timer B on a global level or per device. This makes it
1161 possible to force timeout faster on non-responsive SIP servers. These settings are
1162 considered advanced, so don't use them unless you have a problem.
1163 * Added a dial string option to be able to set the To: header in an INVITE to any
1165 * Added a new global and per-peer option, qualifyfreq, which allows you to configure
1166 the qualify frequency.
1167 * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
1168 were not properly torn down due to network or endpoint failures during an established
1170 * Added experimental TCP and TLS support for SIP. See doc/siptls.txt and
1171 configs/sip.conf.sample for more information on how it is used.
1172 * Added a new configuration option "authfailureevents" that enables manager events when
1173 a peer can't authenticate properly.
1174 * Added DNS manager support to registrations for peers not referencing a peer entry.
1178 * Added the trunkmaxsize configuration option to chan_iax2.
1179 * Added the srvlookup option to iax.conf
1180 * Added support for OSP. The token is set and retrieved through the CHANNEL()
1183 XMPP Google Talk/Jingle changes
1184 -------------------------------
1185 * Added the bindaddr option to gtalk.conf.
1189 * Added skinny show device, skinny show line, and skinny show settings CLI commands.
1190 * Proper codec support in chan_skinny.
1191 * Added settings for IP and Ethernet QoS requests
1195 * Added separate settings for media QoS in mgcp.conf
1197 Console Channel Driver changes
1198 ------------------------------
1199 * Added experimental support for video send & receive to chan_oss.
1200 This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
1203 Phone channel changes (chan_phone)
1204 ----------------------------------
1205 * Added G729 passthrough support to chan_phone for Sigma Designs boards.
1207 H.323 channel Changes
1208 ---------------------
1209 * H323 remote hold notification support added (by NOTIFY message
1210 and/or H.450 supplementary service)
1212 Local channel changes
1213 ---------------------
1214 * The device state functionality in the Local channel driver has been updated
1215 to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
1216 to just UNKNOWN if the extension exists.
1217 * Added jitterbuffer support for chan_local. This allows you to use the
1218 generic jitterbuffer on incoming calls going to Asterisk applications.
1219 For example, this would allow you to use a jitterbuffer for an incoming
1220 SIP call to Voicemail by putting a Local channel in the middle. This
1221 feature is enabled by using the 'j' option in the Dial string to the Local
1222 channel in conjunction with the existing 'n' option for local channels.
1223 * A 'b' option has been added which causes chan_local to return the actual channel
1224 that is behind it when queried. This is useful for transfer scenarios as the
1225 actual channel will be transferred, not the Local channel.
1227 Agent channel changes
1228 ----------------------
1229 * The ackcall and endcall options are now supplemented with options acceptdtmf
1230 and enddtmf. These allow for the DTMF keypress to be configurable. The options
1231 default to their old hard-coded values ('#' and '*' respectively) so this should
1232 not break any existing agent installations.
1234 DAHDI channel driver (chan_dahdi) Changes
1235 ----------------------------------------
1236 * SS7 support (via libss7 library)
1237 * In India, some carriers transmit CID via dtmf. Some code has been added
1238 that will handle some situations. The cidstart=polarity_IN choice has been added for
1239 those carriers that transmit CID via dtmf after a polarity change.
1240 * CID matching information is now shown when doing 'dialplan show'.
1241 * Added dahdi show version CLI command.
1242 * Added setvar support to chan_dahdi.conf channel entries.
1243 * Added two new options: mwimonitor and mwimonitornotify. These options allow
1244 you to enable MWI monitoring on FXO lines. When the MWI state changes,
1245 the script specified in the mwimonitornotify option is executed. An internal
1246 event indicating the new state of the mailbox is also generated, so that
1247 the normal MWI facilities in Asterisk work as usual.
1248 * Added signalling type 'auto', which attempts to use the same signalling type
1249 for a channel as configured in DAHDI. This is primarily designed for analog
1250 ports, but will also work for digital ports that are configured for FXS or FXO
1251 signalling types. This mode is also the default now, so if your chan_dahdi.conf
1252 does not specify signalling for a channel (which is unlikely as the sample
1253 configuration file has always recommended specifying it for every channel) then
1254 the 'auto' mode will be used for that channel if possible.
1255 * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
1256 state for a channel; also ensured that the DNDState Manager event is
1257 emitted no matter how the DND state is set or cleared.
1261 * Added a new channel driver, chan_unistim. See doc/unistim.txt and
1262 configs/unistim.conf.sample for details. This new channel driver allows
1263 you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
1264 * Added a new channel driver, chan_console, which uses portaudio as a cross
1265 platform audio interface. It was written as a channel driver that would
1266 work with Mac CoreAudio, but portaudio supports a number of other audio
1267 interfaces, as well. Note that this channel driver requires v19 or higher
1268 of portaudio; older versions have a different API.
1272 * Added the ability to specify arguments to the Dial application when using
1273 the DUNDi switch in the dialplan.
1274 * Added the ability to set weights for responses dynamically. This can be
1275 done using a global variable or a dialplan function. Using the SHELL()
1276 function would allow you to have an external script set the weight for
1278 * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
1279 functions will allow you to initiate a DUNDi query from the dialplan,
1280 find out how many results there are, and access each one.
1281 * Added the ability to specifiy a port for a dundi peer.
1285 * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
1286 functions will allow you to initiate an ENUM lookup from the dialplan,
1287 and Asterisk will cache the results. ENUMRESULT can be used to access
1288 the results without doing multiple DNS queries.
1292 * Added the ability to customize which sound files are used for some of the
1293 prompts within the Voicemail application by changing them in voicemail.conf
1294 * Added the ability for the "voicemail show users" CLI command to show users
1295 configured by the dynamic realtime configuration method.
1296 * MWI (Message Waiting Indication) handling has been significantly
1297 restructured internally to Asterisk. It is now totally event based
1298 instead of polling based. The voicemail application will notify other
1299 modules that have subscribed to MWI events when something in the mailbox
1301 This also means that if any other entity outside of Asterisk is changing
1302 the contents of mailboxes, then the voicemail application still needs to
1303 poll for changes. Examples of situations that would require this option
1304 are web interfaces to voicemail or an email client in the case of using
1305 IMAP storage. So, two new options have been added to voicemail.conf
1306 to account for this: "pollmailboxes" and "pollfreq". See the sample
1307 configuration file for details.
1308 * Added "tw" language support
1309 * Added support for storage of greetings using an IMAP server
1310 * Added ability to customize forward, reverse, stop, and pause keys for message playback
1311 * SMDI is now enabled in voicemail using the smdienable option.
1312 * A "lockmode" option has been added to asterisk.conf to configure the file
1313 locking method used for voicemail, and potentially other things in the
1314 future. The default is the old behavior, lockfile. However, there is a
1315 new method, "flock", that uses a different method for situations where the
1316 lockfile will not work, such as on SMB/CIFS mounts.
1317 * Added the ability to backup deleted messages, to ease recovery in the case
1318 that a user accidentally deletes a message, and discovers that they need it.
1319 * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
1320 is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
1321 smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
1322 voicemail boxes. The SMDI interface can also poll for MWI changes when some
1323 outside entity is modifying the state of the mailbox (such as IMAP storage or
1324 a web interface of some kind).
1325 * Added the support for marking messages as "urgent." There are two methods to accomplish
1326 this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
1327 is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
1328 the message as urgent after he has recorded a voicemail by following the voice instructions.
1329 When listening to voicemails using VoiceMailMain urgent messages will be presented before other
1334 * Added the general option 'shared_lastcall' so that member's wrapuptime may be
1335 used across multiple queues.
1336 * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
1337 setqueueentryvar options for each queue, see queues.conf.sample for details.
1338 * Added keepstats option to queues.conf which will keep queue
1339 statistics during a reload.
1340 * setinterfacevar option in queues.conf also now sets a variable
1341 called MEMBERNAME which contains the member's name.
1342 * Added 'Strategy' field to manager event QueueParams which represents
1343 the queue strategy in use.
1344 * Added option to run macro when a queue member is connected to a caller,
1345 see queues.conf.sample for details.
1346 * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
1347 does not count paused queue members as unavailable.
1348 * Added min-announce-frequency option to queues.conf which allows you to control the
1349 minimum amount of time between queue announcements for use when the caller's queue
1350 position changes frequently.
1351 * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
1353 * Added ability for non-realtime queues to have realtime members
1354 * Added the "linear" strategy to queues.
1355 * Added the "wrandom" strategy to queues.
1356 * Added new channel variable QUEUE_MIN_PENALTY
1357 * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
1358 rules in queuerules.conf. See configs/queuerules.conf.sample for details
1359 * Added a new parameter for member definition, called state_interface. This may be
1360 used so that a member may be called via one interface but have a different interface's
1361 device state reported.
1362 * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
1363 "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
1364 "manager show command QueueReset."
1365 * New configuration option: randomperiodicannounce. If a list of periodic announcements is
1366 specified by the periodic-announce option, then one will be chosen randomly when it is time
1367 to play a periodic announcment
1368 * New configuration options: announce-position now takes two more values in addition to "yes" and
1369 "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
1370 announce-position-limit. By setting announce-position to "limit" callers will only have their
1371 position announced if their position is less than what is specified by announce-position-limit.
1372 If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
1373 will be told that their are more than announce-position-limit callers waiting.
1374 * Two new queue log events have been added. An ADDMEMBER event will be logged
1375 when a realtime queue member is added and a REMOVEMEMBER event will be logged
1376 when a realtime queue member is removed. Since there is no calling channel associated
1377 with these events, the string "REALTIME" is placed where the channel's unique id
1378 is typically placed.
1379 * The configuration method for the "joinempty" and "leavewhenempty" options has
1380 changed to a comma-separated list of methods of determining member availability
1381 instead of vague terms such as "yes," "loose," "no," and "strict." These old four
1382 values are still accepted for backwards-compatibility, though.
1383 * The average talktime is now calculated on queues. This information is reported via the
1384 CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
1385 and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
1390 * The 'o' option to provide an optimization has been removed and its functionality
1391 has been enabled by default.
1392 * When a conference is created, the UNIQUEID of the channel that caused it to be
1393 created is stored. Then, every channel that joins the conference will have the
1394 MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
1395 callers that come and go from long standing conferences.
1396 * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
1397 except it does operations on a channel by name, instead of number in a conference.
1398 This is a very useful feature in combination with the 'X' option to ChanSpy.
1399 * Added 'C' option to Meetme which causes a caller to continue in the dialplan
1401 * Added new RealTime functionality to provide support for scheduled conferencing.
1402 This includes optional messages to the caller if they attempt to join before
1403 the schedule start time, or to allow the caller to join the conference early.
1404 Also included is optional support for limiting the number of callers per
1405 RealTime conference.
1406 * Added the S() and L() options to the MeetMe application. These are pretty
1407 much identical to the S() and L() options to Dial(). They let you set
1408 timeouts for the conference, as well as have warning sounds played to
1409 let the caller know how much time is left, and when it is running out.
1410 * Added the ability to do "meetme concise" with the "meetme" CLI command.
1411 This extends the concise capabilities of this CLI command to include
1412 listing all conferences, instead of an addition to the other sub commands
1413 for the "meetme" command.
1414 * Added the ability to specify the music on hold class used to play into the
1415 conference when there is only one member and the M option is used.
1416 * Added MEETME_INFO dialplan function which provides a way to query
1417 various properties of a Meetme conference.
1418 * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
1419 and *84: record in-conf
1421 Other Dialplan Application Changes
1422 ----------------------------------
1423 * Argument support for Gosub application
1424 * From the to-do lists: straighten out the app timeout args:
1425 Wait() app now really does 0.3 seconds- was truncating arg to an int.
1426 WaitExten() same as Wait().
1427 Congestion() - Now takes floating pt. argument.
1428 Busy() - now takes floating pt. argument.
1429 Read() - timeout now can be floating pt.
1430 WaitForRing() now takes floating pt timeout arg.
1431 SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
1432 * Added 's' option to Page application.
1433 * Added an optional timeout argument to the Page application.
1434 * Added 'E', 'V', and 'P' commands to ExternalIVR.
1435 * Added 'o' and 'X' options to Chanspy.
1436 * Added a new dialplan application, Bridge, which allows you to bridge the
1437 calling channel to any other active channel on the system.
1438 * Added the ability to specify a music on hold class to play instead of ringing
1439 for the SLATrunk application.
1440 * The Read application no longer exits the dialplan on error. Instead, it sets
1441 READSTATUS to ERROR, which you can catch and handle separately.
1442 * Added 'm' option to Directory, which lists out names, 8 at a time, instead
1443 of asking for verification of each name, one at a time.
1444 * Privacy() no longer uses privacy.conf, as all options are specifyable as
1445 direct options to the app.
1446 * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
1448 * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
1449 * The ChannelRedirect application no longer exits the dialplan if the given channel
1450 does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
1451 or NOCHANNEL if the given channel was not found.
1452 * The silencethreshold setting that was previously configurable in multiple
1453 applications is now settable globally via dsp.conf.
1455 Music On Hold Changes
1456 ---------------------
1457 * A new option, "digit", has been added for music on hold classes in
1458 musiconhold.conf. If this is set for a music on hold class, a caller
1459 listening to music on hold can press this digit to switch to listening
1460 to this music on hold class.
1461 * Support for realtime music on hold has been added.
1462 * In conjunction with the realtime music on hold, a general section has
1463 been added to musiconhold.conf, its sole variable is cachertclasses. If this
1464 is set, then music on hold classes found in realtime will be cached in memory.
1468 * AEL upgraded to use the Gosub with Arguments instead
1469 of Macro application, to hopefully reduce the problems
1470 seen with the artificially low stack ceiling that
1471 Macro bumps into. Macros can only call other Macros
1472 to a depth of 7. Tests run using gosub, show depths
1473 limited only by virtual memory. A small test demonstrated
1474 recursive call depths of 100,000 without problems.
1475 -- in addition to this, all apps that allowed a macro
1476 to be called, as in Dial, queues, etc, are now allowing
1477 a gosub call in similar fashion.
1478 * AEL now generates LOCAL(argname) declarations when it
1479 Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
1480 etc. That makes the arguments local in scope. The user
1481 can define their own local variables in macros, now,
1482 by saying "local myvar=someval;" or using Set() in this
1483 fashion: Set(LOCAL(myvar)=someval); ("local" is now
1485 * utils/conf2ael introduced. Will convert an extensions.conf
1486 file into extensions.ael. Very crude and unfinished, but
1487 will be improved as time goes by. Should be useful for a
1488 first pass at conversion.
1489 * aelparse will now read extensions.conf to see if a referenced
1490 macro or context is there before issueing a warning.
1491 * AEL parser sets a local channel variable ~~EXTEN~~, to
1492 preserve the value of ${EXTEN} thru switch statements.
1493 * New operator in $[...] expressions: the ~~ operator serves
1494 as a concatenation operator. AT THE MOMENT, it is really only
1495 necessary and useful in AEL, especially in if() expressions.
1496 Operation: ${a} ~~ ${b| with force both a and b to strings, strip
1497 any enclosing double-quotes, and evaluate to the value of a
1498 concatenated with the value of b. For example if a is set to
1499 "xyz" and b has the value "abc", then ${a} ~~ ${b| would
1500 evaluate to xyzabc .
1503 Call Features (res_features) Changes
1504 ------------------------------------
1505 * Added the parkedcalltransfers option to features.conf
1506 * Added parkedcallparking option to control one touch parking w/ parking
1508 * Added parkedcallhangup option to control disconnect feature w/ parking
1510 * Added parkedcallrecording option to control one-touch record w/ parking
1512 * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
1513 parkedcalltransfers option support for multiple parking lots.
1514 * Added BRIDGE_FEATURES variable to set available features for a channel
1515 * The built-in method for doing attended transfers has been updated to
1516 include some new options that allow you to have the transferee sent
1517 back to the person that did the transfer if the transfer is not successful.
1518 See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
1519 in features.conf.sample.
1520 * Added support for configuring named groups of custom call features in
1521 features.conf. This means that features can be written a single time, and
1522 then mapped into groups of features for different key mappings or easier
1524 * Updated the ParkedCall application to allow you to not specify a parking
1525 extension. If you don't specify a parking space to pick up, it will grab
1526 the first one available.
1527 * Added cli command 'features reload' to reload call features from features.conf
1528 * Moved into core asterisk binary.
1529 * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
1530 * Added the ability for custom parking lots to be configured with their own
1531 parking extension with the parkext option.
1533 Language Support Changes
1534 ------------------------
1535 * Brazilian Portuguese (pt-BR) in VM, and say.c was added
1536 * Added support for the Hungarian language for saying numbers, dates, and times.
1540 * Added SPEECH commands for speech recognition. A complete listing can be found
1542 * If app_stack is loaded, GOSUB is a native AGI command that may be used to
1543 invoke subroutines in the dialplan. Note that calling EXEC with Gosub
1544 does not behave as expected; the native command needs to be used, instead.
1545 * Added the ability to perform SRV lookups on fast AGI calls. To use this
1546 feature, simply use hagi: instead of agi: as the protocol portion
1547 of the URI parameter to the AGI function call in your dial plan. Also note
1548 that specifying a port number in the AGI URI will disable SRV lookups,
1549 even if you use the hagi: protocol.
1550 * No longer support MSG_OOB flag on HANGUP.
1554 * Added rotatestrategy option to logger.conf, along with two new options:
1555 "timestamp" which will use the time to name the logger files instead of
1556 sequence number; and "rotate", which rotates the names of the log files,
1557 similar to the way syslog rotates files.
1558 * Added exec_after_rotate option to logger.conf, which allows a system
1559 command to be run after rotation. This is primarily useful with
1560 rotatestrategy=rotate, to allow a limit on the number of log files kept
1561 and to ensure that the oldest log file gets deleted.
1562 * Added realtime support for the queue log
1566 * The cdr_manager module has a [mappings] feature, like cdr_custom,
1567 to add fields to the manager event from the CDR variables.
1568 * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
1569 backend database CDR table. Specifically, additional, non-standard
1570 columns are supported, merely by setting the corresponding CDR variable in
1571 your dialplan. In addition, you may alias any column to another name (for
1572 example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
1573 simply "alias src => ANI" in the configuration file). Records may be
1574 posted to more than one backend, simply by specifying multiple categories
1575 in the configuration file. And finally, you may filter which CDRs get
1576 posted to each backend, by specifying a filter (which the record must
1577 match) for the particular category. Filters are additive (meaning all
1578 rules must match to post that CDR).
1579 * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
1580 module. Specifically, you may add additional columns into the table and
1581 they will be set, if you set the corresponding CDR variable name. Also,
1582 if you omit columns in your database table, they will be silently skipped
1583 (but a record will still be inserted, based on what columns remain). Note
1584 that the other two features from cdr_adaptive_odbc (alias and filter) are
1585 not currently supported.
1586 * The ResetCDR application now has an 'e' option that re-enables a CDR if it
1587 has been disabled using the NoCDR application.
1589 Miscellaneous New Modules
1590 -------------------------
1591 * Added a new CDR module, cdr_sqlite3_custom.
1592 * Added a new realtime configuration module, res_config_sqlite
1593 * Added a new codec translation module, codec_resample, which re-samples
1594 signed linear audio between 8 kHz and 16 kHz to help support wideband
1596 * Added a new module, res_phoneprov, which allows auto-provisioning of phones
1597 based on configuration templates that use Asterisk dialplan function and
1598 variable substitution. It should be possible to create phone profiles and
1599 templates that work for the majority of phones provisioned over http. It
1600 is currently only intended to provision a single user account per phone.
1601 An example profile and set of templates for Polycom phones is provided.
1602 NOTE: Polycom firmware is not included, but should be placed in
1603 AST_DATA_DIR/phoneprov/configs to match up with the included templates.
1604 * Added a new module, app_jack, which provides interfaces to JACK, the Jack
1605 Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
1606 provided; there is a JACK() application, and a JACK_HOOK() function. Both
1607 interfaces create an input and output JACK port. The application makes
1608 these ports the endpoint of the call. The audio coming from the channel
1609 goes out the output port and whatever comes back in on the input port is
1610 what gets sent to the channel. The JACK_HOOK() function turns on a JACK
1611 audiohook on the channel. This lets you run the audio coming from a
1612 channel through JACK, and whatever comes back in is what gets forwarded
1613 on as the channel's audio. This is very useful for building custom
1614 vocoders or doing recording or analysis of the channel's audio in another
1616 * Added a new module, res_config_curl, which permits using a HTTP POST url
1617 to retrieve, create, update, and delete realtime information from a remote
1618 web server. Note that this module requires func_curl.so to be loaded for
1619 backend functionality.
1620 * Added a new module, res_config_ldap, which permits the use of an LDAP
1621 server for realtime data access.
1622 * Added support for writing and running your dialplan in lua using the pbx_lua
1623 module. See configs/extensions.lua.sample for examples of how to do this.
1627 * Ability to use libcap to set high ToS bits when non-root
1628 on Linux. If configure is unable to find libcap then you
1629 can use --with-cap to specify the path.
1630 * Added maxfiles option to options section of asterisk.conf which allows you to specify
1631 what Asterisk should set as the maximum number of open files when it loads.
1632 * Added the jittertargetextra configuration option.
1633 * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
1634 configuration files for the IP channel drivers. The new option is "cos".
1635 This information is also documented in doc/qos.tex, or the IP Quality of Service
1636 section of asterisk.pdf.
1637 * When originating a call using AMI or pbx_spool that fails the reason for failure
1638 will now be available in the failed extension using the REASON dialplan variable.
1639 * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
1640 It allows you to configure a prefix for auto-monitor recordings.
1641 * A new extension pattern matching algorithm, based on a trie, is introduced
1642 here, that could noticeably speed up mid-sized to large dialplans.
1643 It is NOT used by default, as duplicating the behaviour of the old pattern
1644 matcher is still under development. A config file option, in extensions.conf,
1645 in the [general] section, called "extenpatternmatchingnew", is by default
1646 set to false; setting that to true will force the use of the new algorithm.
1647 Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
1648 be used to switch the algorithms at run time.
1649 * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
1650 specifying which socket to use to connect to the running Asterisk daemon
1652 * Performance enhancements to the sched facility, which is used in
1653 the channel drivers, etc. Added hashtabs and doubly-linked lists
1654 to speed up deletion; start at the beginning or end of list to
1656 * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
1657 dlinkedlists.h. Doubly-linked lists feature fast deletion times.
1658 Added regression tests to the tests/ dir, also.
1659 * Added a refcount trace feature to astobj2 for those trying to balance
1660 object creation, deletion; work, play; space and time. See the
1661 notes in astobj2.h. Also, see utils/refcounter as well, as a
1662 quick way to find unbalanced refcounts in what could be a sea
1663 of objects that were balanced.
1664 * Added logging to 'make update' command. See update.log
1665 * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
1666 do not come from the remote party.
1667 * Added the 'n' option to the SpeechBackground application to tell it to not
1668 answer the channel if it has not already been answered.
1669 * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
1670 turned on, via the CHANNEL(trace) dialplan function. Could be useful for
1672 * iLBC source code no longer included (see UPGRADE.txt for details)
1673 * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
1674 deadlock is detected, a backtrace of the stack which led to the lock calls
1675 will be output to the CLI.
1676 * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
1677 the "core show locks" CLI command will give lock information output as well
1678 as a backtrace of the stack which led to the lock calls.
1679 * users.conf now sports an optional alternateexts property, which permits
1680 allocation of additional extensions which will reach the specified user.
1681 * A new option for the configure script, --enable-internal-poll, has been added
1682 for use with systems which may have a buggy implementation of the poll system
1683 call. If you notice odd behavior such as the CLI being unresponsive on remote
1684 consoles, you may want to try using this option. This option is enabled by default
1685 on Darwin systems since it is known that the Darwin poll() implementation has
1689 --------------------
1690 * In addition to timing from DAHDI, there is a new timing module called
1691 res_timing_timerfd. In order to use this, you must be running Linux with
1692 a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
1693 script will be able to tell if you have the requirements. From menuselect, select
1694 res_timing_timerfd from the Resource Modules menu.
projects / asterisk/asterisk.git / blob