------------------------------------------------------------------------------- --- Functionality changes since Asterisk 1.4-beta was branched ---------------- ------------------------------------------------------------------------------- AMI - The manager (TCP/TLS/HTTP) -------------------------------- * Added the URI redirect option for the built-in HTTP server * The output of CallerID in Manager events is now more consistent. CallerIDNum is used for number and CallerIDName for name. * enable https support for builtin web server. See configs/http.conf.sample for details. * Added a new action, GetConfigJSON, which can return the contents of an Asterisk configuration file in JSON format. This is intended to help improve the performance of AJAX applications using the manager interface over HTTP. * SIP and IAX manager events now use "ChannelType" in all cases where we indicate channel driver. Previously, we used a mixture of "Channel" and "ChannelDriver" headers. * Added a "Bridge" action which allows you to bridge any two channels that are currently active on the system. Dialplan functions ------------------ * Added the DEVSTATE() dialplan function which allows retrieving any device state in the dialplan, as well as creating custom device states that are controllable from the dialplan. * Extend CALLERID() function with "pres" and "ton" parameters to fetch string representation of calling number presentation indicator and numeric representation of type of calling number value. * MailboxExists converted to dialplan function CLI Changes ----------- * New CLI command "core show settings" * Added 'core show channels count' CLI command. SIP changes ----------- * The default SIP useragent= identifier now includes the Asterisk version * A new option, match_auth_username in sip.conf changes the matching of incoming requests. If set, and the incoming request carries authentication info, the username to match in the users list is taken from the Digest header rather than from the From: field. This feature is considered experimental. * The "musiconhold" and "musicclass" settings in sip.conf are now removed, since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4 * The "localmask" setting was removed in version 1.2 and the reminder about it being removed is now also removed. * A new option "busy-level" for setting a level of calls where asterisk reports a device as busy, to separate it from call-limit * A new realtime family called "sipregs" is now supported to store SIP registration data. If this family is defined, "sippeers" will be used for configuration and "sipregs" for registrations. If it's not defined, "sippeers" will be used for registration data, as before. * The SIPPEER function have new options for port address, call and pickup groups * Added support for T.140 realtime text in SIP/RTP * The "checkmwi" option has been removed from sip.conf, as it is no longer required due to the restructuring of how MWI is handled. See the descriptions in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf for more information. IAX2 changes ------------ * Added the trunkmaxsize configuration option to chan_iax2. * Added the srvlookup option to iax.conf * Added support for OSP. The token is set and retrieved through the CHANNEL() dialplan function. DUNDi changes ------------- * Added the ability to specify arguments to the Dial application when using the DUNDi switch in the dialplan. * Added the ability to set weights for responses dynamically. This can be done using a global variable or a dialplan function. Using the SHELL() function would allow you to have an external script set the weight for each response. * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These functions will allow you to initiate a DUNDi query from the dialplan, find out how many results there are, and access each one. Voicemail Changes ----------------- * Added the ability to customize which sound files are used for some of the prompts within the Voicemail application by changing them in voicemail.conf * Added the ability for the "voicemail show users" CLI command to show users configured by the dynamic realtime configuration method. * MWI (Message Waiting Indication) handling has been significantly restructured internally to Asterisk. It is now totally event based instead of polling based. The voicemail application will notify other modules that have subscribed to MWI events when something in the mailbox changes. This also means that if any other entity outside of Asterisk is changing the contents of mailboxes, then the voicemail application still needs to poll for changes. Examples of situations that would require this option are web interfaces to voicemail or an email client in the case of using IMAP storage. So, two new options have been added to voicemail.conf to account for this: "pollmailboxes" and "pollfreq". See the sample configuration file for details. Queue changes ------------- * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and setqueueentryvar options for each queue, see queues.conf.sample for details. * Added keepstats option to queues.conf which will keep queue statistics during a reload. * setinterfacevar option in queues.conf also now sets a variable called MEMBERNAME which contains the member's name. * Added 'Strategy' field to manager event QueueParams which represents the queue strategy in use. * Added option to run macro when a queue member is connected to a caller, see queues.conf.sample for details. * app_queue now has a 'loose' option which is almost exactly like 'strict' except it does not count paused queue members as unavailable. * Added min-announce-frequency option to queues.conf which allows you to control the minimum amount of time between queue announcements for use when the caller's queue position changes frequently. * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the queue log. MeetMe Changes -------------- * The 'o' option to provide an optimization has been removed and its functionality has been enabled by default. * When a conference is created, the UNIQUEID of the channel that caused it to be created is stored. Then, every channel that joins the conference will have the MEETMEUNIQUEID channel variable set with this ID. This can be used to relate callers that come and go from long standing conferences. * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin, except it does operations on a channel by name, instead of number in a conference. This is a very useful feature in combination with the 'X' option to ChanSpy. Miscellaneous ------------- * Added the bindaddr option to gtalk.conf. * Argument support for Gosub application * Ability to set process limits without restarting Asterisk * SS7 support in chan_zap (via libss7 library) * Proper codec support in chan_skinny. * AEL upgraded to use the Gosub with Arguments instead of Macro application, to hopefully reduce the problems seen with the artificially low stack ceiling that Macro bumps into. Macros can only call other Macros to a depth of 7. Tests run using gosub, show depths limited only by virtual memory. A small test demonstrated recursive call depths of 100,000 without problems. * Ability to use libcap to set high ToS bits when non-root on Linux. If configure is unable to find libcap then you can use --with-cap to specify the path. * H323 remote hold notification support added (by NOTIFY message and/or H.450 supplementary service) * Added rotatetimestamp option to logger.conf which will use the time to name the logger files instead of sequence number. * Added Masquerade manager event for when a masquerade happens between two channels. * From the to-do lists: straighten out the app timeout args: Wait() app now really does 0.3 seconds- was truncating arg to an int. WaitExten() same as Wait(). Congestion() - Now takes floating pt. argument. Busy() - now takes floating pt. argument. Read() - timeout now can be floating pt. WaitForRing() now takes floating pt timeout arg. SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds. * Added 'C' option to Meetme which causes a caller to continue in the dialplan when kicked out. * Brazilian Portuguese (pt-BR) in VM, and say.c was added via patch from cfassoni. * CID matching information is now shown when doing 'dialplan show'. * Added maxfiles option to options section of asterisk.conf which allows you to specify what Asterisk should set as the maximum number of open files when it loads. * Added the jittertargetextra configuration option. * Added G729 passthrough support to chan_phone for Sigma Designs boards. * Added the parkedcalltransfers option to features.conf * Added 's' option to Page application. * Added 'E' and 'V' commands to ExternalIVR. * Added 'DBDel' and 'DBDelTree' manager commands. * Added 'o' and 'X' options to Chanspy. * Added the parkedcallreparking option to features.conf * SMDI is now enabled in voicemail using the smdienable option. * Added zap show version CLI command to chan_zap. * Added a new CDR module, cdr_sqlite3_custom. * Added a new realtime configuration module, res_config_sqlite * Added a new dialplan application, Bridge, which allows you to bridge the calling channel to any other active channel on the system. * Added support for setting the CoS for VLAN traffic (802.1p). See the sample configuration files for the IP channel drivers. The new option is "cos". This information is also documented in doc/qos.tex, or the IP Quality of Service section of asterisk.pdf. * The built-in method for doing attended transfers has been updated to include some new options that allow you to have the transferee sent back to the person that did the transfer if the transfer is not successful. See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries" in features.conf.sample. * The device state functionality in the Local channel driver has been updated to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed to just UNKNOWN if the extension exists. * Added support for the Hungarian language for saying numbers, dates, and times.