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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
305 lines
17 KiB
Plaintext
305 lines
17 KiB
Plaintext
------------------------------------------------------------------------------
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--- Functionality changes since Asterisk 1.4-beta was branched ----------------
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-------------------------------------------------------------------------------
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AMI - The manager (TCP/TLS/HTTP)
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--------------------------------
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* Added TLS support for the manager interface and HTTP server
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* Added the URI redirect option for the built-in HTTP server
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* The output of CallerID in Manager events is now more consistent.
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CallerIDNum is used for number and CallerIDName for name.
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* enable https support for builtin web server.
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See configs/http.conf.sample for details.
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* Added a new action, GetConfigJSON, which can return the contents of an
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Asterisk configuration file in JSON format. This is intended to help
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improve the performance of AJAX applications using the manager interface
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over HTTP.
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* SIP and IAX manager events now use "ChannelType" in all cases where we
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indicate channel driver. Previously, we used a mixture of "Channel"
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and "ChannelDriver" headers.
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* Added a "Bridge" action which allows you to bridge any two channels that
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are currently active on the system.
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* Added a "ListAllVoicemailUsers" action that allows you to get a list of all
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the voicemail users setup.
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* Added 'DBDel' and 'DBDelTree' manager commands.
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Dialplan functions
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------------------
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* Added the DEVICE_STATE() dialplan function which allows retrieving any device
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state in the dialplan, as well as creating custom device states that are
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controllable from the dialplan.
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* Extend CALLERID() function with "pres" and "ton" parameters to
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fetch string representation of calling number presentation indicator
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and numeric representation of type of calling number value.
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* MailboxExists converted to dialplan function
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* A new option to Dial() for telling IP phones not to count the call
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as "missed" when dial times out and cancels.
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* Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
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mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
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held for any given channel. Also, locks are automatically freed when a
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channel is hung up.
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* Added HINT() dialplan function that allows retrieving hint information.
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Hints are mappings between extensions and devices for the sake of
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determining the state of an extension. This function can retrieve the list
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of devices or the name associated with a hint.
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* Added EXTENSION_STATE() dialplan function which allows retrieving the state
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of any extension.
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CLI Changes
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-----------
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* New CLI command "core show settings"
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* Added 'core show channels count' CLI command.
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* Added the ability to set the core debug and verbose values on a per-file basis.
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* Added 'queue pause member' and 'queue unpause member' CLI commands
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SIP changes
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-----------
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* Improved NAT and STUN support.
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chan_sip now can use port numbers in bindaddr, externip and externhost
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options, as well as contact a STUN server to detect its external address
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for the SIP socket. See sip.conf.sample, 'NAT' section.
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* The default SIP useragent= identifier now includes the Asterisk version
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* A new option, match_auth_username in sip.conf changes the matching of incoming requests.
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If set, and the incoming request carries authentication info,
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the username to match in the users list is taken from the Digest header
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rather than from the From: field. This feature is considered experimental.
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* The "musiconhold" and "musicclass" settings in sip.conf are now removed,
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since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
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* The "localmask" setting was removed in version 1.2 and the reminder about it
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being removed is now also removed.
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* A new option "busy-level" for setting a level of calls where asterisk reports
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a device as busy, to separate it from call-limit
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* A new realtime family called "sipregs" is now supported to store SIP registration
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data. If this family is defined, "sippeers" will be used for configuration and
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"sipregs" for registrations. If it's not defined, "sippeers" will be used for
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registration data, as before.
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* The SIPPEER function have new options for port address, call and pickup groups
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* Added support for T.140 realtime text in SIP/RTP
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* The "checkmwi" option has been removed from sip.conf, as it is no longer
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required due to the restructuring of how MWI is handled. See the descriptions
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in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
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for more information.
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* Added rtpdest option to CHANNEL() dialplan function.
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* Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
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* SIP now adds a header to the CANCEL if the call was answered by another phone
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in the same dial command, or if the new c option in dial() is used.
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* The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
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states it is not needed. For phones, however, that do require it the registertrying option
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has been added so it can be enabled.
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IAX2 changes
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------------
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* Added the trunkmaxsize configuration option to chan_iax2.
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* Added the srvlookup option to iax.conf
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* Added support for OSP. The token is set and retrieved through the CHANNEL()
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dialplan function.
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Skinny changes
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-------------
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* Added skinny show device, skinny show line, and skinny show settings CLI commands.
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DUNDi changes
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-------------
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* Added the ability to specify arguments to the Dial application when using
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the DUNDi switch in the dialplan.
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* Added the ability to set weights for responses dynamically. This can be
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done using a global variable or a dialplan function. Using the SHELL()
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function would allow you to have an external script set the weight for
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each response.
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* Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
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functions will allow you to initiate a DUNDi query from the dialplan,
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find out how many results there are, and access each one.
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ENUM changes
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------------
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* Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
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functions will allow you to initiate an ENUM lookup from the dialplan,
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and Asterisk will cache the results. ENUMRESULT can be used to access
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the results without doing multiple DNS queries.
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Voicemail Changes
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-----------------
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* Added the ability to customize which sound files are used for some of the
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prompts within the Voicemail application by changing them in voicemail.conf
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* Added the ability for the "voicemail show users" CLI command to show users
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configured by the dynamic realtime configuration method.
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* MWI (Message Waiting Indication) handling has been significantly
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restructured internally to Asterisk. It is now totally event based
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instead of polling based. The voicemail application will notify other
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modules that have subscribed to MWI events when something in the mailbox
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changes.
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This also means that if any other entity outside of Asterisk is changing
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the contents of mailboxes, then the voicemail application still needs to
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poll for changes. Examples of situations that would require this option
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are web interfaces to voicemail or an email client in the case of using
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IMAP storage. So, two new options have been added to voicemail.conf
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to account for this: "pollmailboxes" and "pollfreq". See the sample
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configuration file for details.
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* Added "tw" language support
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* Added support for storage of greetings using an IMAP server
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* Added ability to customize forward, reverse, stop, and pause keys for message playback
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* SMDI is now enabled in voicemail using the smdienable option.
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* A "lockmode" option has been added to asterisk.conf to configure the file
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locking method used for voicemail, and potentially other things in the
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future. The default is the old behavior, lockfile. However, there is a
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new method, "flock", that uses a different method for situations where the
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lockfile will not work, such as on SMB/CIFS mounts.
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Queue changes
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-------------
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* Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
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setqueueentryvar options for each queue, see queues.conf.sample for details.
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* Added keepstats option to queues.conf which will keep queue
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statistics during a reload.
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* setinterfacevar option in queues.conf also now sets a variable
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called MEMBERNAME which contains the member's name.
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* Added 'Strategy' field to manager event QueueParams which represents
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the queue strategy in use.
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* Added option to run macro when a queue member is connected to a caller,
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see queues.conf.sample for details.
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* app_queue now has a 'loose' option which is almost exactly like 'strict' except it
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does not count paused queue members as unavailable.
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* Added min-announce-frequency option to queues.conf which allows you to control the
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minimum amount of time between queue announcements for use when the caller's queue
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position changes frequently.
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* Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
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queue log.
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* Added ability for non-realtime queues to have realtime members
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MeetMe Changes
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--------------
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* The 'o' option to provide an optimization has been removed and its functionality
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has been enabled by default.
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* When a conference is created, the UNIQUEID of the channel that caused it to be
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created is stored. Then, every channel that joins the conference will have the
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MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
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callers that come and go from long standing conferences.
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* Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
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except it does operations on a channel by name, instead of number in a conference.
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This is a very useful feature in combination with the 'X' option to ChanSpy.
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* Added 'C' option to Meetme which causes a caller to continue in the dialplan
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when kicked out.
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* Added new RealTime functionality to provide support for scheduled conferencing.
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This includes optional messages to the caller if they attempt to join before
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the schedule start time, or to allow the caller to join the conference early.
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Also included is optional support for limiting the number of callers per
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RealTime conference.
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Music On Hold Changes
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---------------------
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* A new option, "digit", has been added for music on hold classes in
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musiconhold.conf. If this is set for a music on hold class, a caller
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listening to music on hold can press this digit to switch to listening
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to this music on hold class.
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AEL Changes
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-----------
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* AEL upgraded to use the Gosub with Arguments instead
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of Macro application, to hopefully reduce the problems
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seen with the artificially low stack ceiling that
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Macro bumps into. Macros can only call other Macros
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to a depth of 7. Tests run using gosub, show depths
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limited only by virtual memory. A small test demonstrated
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recursive call depths of 100,000 without problems.
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-- in addition to this, all apps that allowed a macro
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to be called, as in Dial, queues, etc, are now allowing
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a gosub call in similar fashion.
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* AEL now generates LOCAL(argname) declarations when it
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Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
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etc. That makes the arguments local in scope. The user
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can define their own local variables in macros, now,
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by saying "local myvar=someval;" or using Set() in this
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fashion: Set(LOCAL(myvar)=someval); ("local" is now
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an AEL keyword).
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* utils/conf2ael introduced. Will convert an extensions.conf
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file into extensions.ael. Very crude and unfinished, but
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will be improved as time goes by. Should be useful for a
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first pass at conversion.
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* aelparse will now read extensions.conf to see if a referenced
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macro or context is there before issueing a warning.
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Zaptel channel driver (chan_zap) Changes
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----------------------------------------
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* SS7 support in chan_zap (via libss7 library)
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* In India, some carriers transmit CID via dtmf. Some code has been added
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that will handle some situations. The cidstart=polarity_IN choice has been added for
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those carriers that transmit CID via dtmf after a polarity change.
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* CID matching information is now shown when doing 'dialplan show'.
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* Added zap show version CLI command to chan_zap.
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* Added setvar support to zapata.conf channel entries.
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H.323 Changes
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-------------
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* H323 remote hold notification support added (by NOTIFY message
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and/or H.450 supplementary service)
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Call Features (res_features) Changes
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------------------------------------
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* Added the parkedcalltransfers option to features.conf
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* The built-in method for doing attended transfers has been updated to
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include some new options that allow you to have the transferee sent
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back to the person that did the transfer if the transfer is not successful.
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See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
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in features.conf.sample.
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* Added support for configuring named groups of custom call features in
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features.conf. This means that features can be written a single time, and
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then mapped into groups of features for different key mappings or easier
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access control.
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Language Support Changes
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------------------------
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* Brazilian Portuguese (pt-BR) in VM, and say.c was added
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* Added support for the Hungarian language for saying numbers, dates, and times.
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Miscellaneous
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-------------
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* Added the bindaddr option to gtalk.conf.
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* Argument support for Gosub application
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* Ability to set process limits without restarting Asterisk
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* Proper codec support in chan_skinny.
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* Ability to use libcap to set high ToS bits when non-root
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on Linux. If configure is unable to find libcap then you
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can use --with-cap to specify the path.
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* Added rotatetimestamp option to logger.conf which will use
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the time to name the logger files instead of sequence number.
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* Added Masquerade manager event for when a masquerade happens between
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two channels.
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* From the to-do lists: straighten out the app timeout args:
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Wait() app now really does 0.3 seconds- was truncating arg to an int.
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WaitExten() same as Wait().
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Congestion() - Now takes floating pt. argument.
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Busy() - now takes floating pt. argument.
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Read() - timeout now can be floating pt.
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WaitForRing() now takes floating pt timeout arg.
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SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
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* Added maxfiles option to options section of asterisk.conf which allows you to specify
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what Asterisk should set as the maximum number of open files when it loads.
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* Added the jittertargetextra configuration option.
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* Added G729 passthrough support to chan_phone for Sigma Designs boards.
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* Added 's' option to Page application.
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* Added 'E' and 'V' commands to ExternalIVR.
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* Added 'o' and 'X' options to Chanspy.
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* Added a new CDR module, cdr_sqlite3_custom.
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* The cdr_manager module has a [mappings] feature, like cdr_custom,
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to add fields to the manager event from the CDR variables.
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* Added a new realtime configuration module, res_config_sqlite
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* Added a new dialplan application, Bridge, which allows you to bridge the
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calling channel to any other active channel on the system.
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* Added support for setting the CoS for VLAN traffic (802.1p). See the sample
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configuration files for the IP channel drivers. The new option is "cos".
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This information is also documented in doc/qos.tex, or the IP Quality of Service
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section of asterisk.pdf.
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* The device state functionality in the Local channel driver has been updated
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to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
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to just UNKNOWN if the extension exists.
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* When originating a call using AMI or pbx_spool that fails the reason for failure
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will now be available in the failed extension using the REASON dialplan variable.
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* Added jitterbuffer support for chan_local. This allows you to use the
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generic jitterbuffer on incoming calls going to Asterisk applications.
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For example, this would allow you to use a jitterbuffer for an incoming
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SIP call to Voicemail by putting a Local channel in the middle. This
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feature is enabled by using the 'j' option in the Dial string to the Local
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channel in conjunction with the existing 'n' option for local channels.
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* Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
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It allows you to configure a prefix for auto-monitor recordings.
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