path: root/CHANGES

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--- Functionality changes since Asterisk 1.4-beta was branched ----------------
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AMI - The manager (TCP/TLS/HTTP)
--------------------------------
  * Added TLS support for the manager interface and HTTP server
  * 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.
  * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
     the voicemail users setup.
  * Added 'DBDel' and 'DBDelTree' manager commands.

Dialplan functions
------------------
  * Added the DEVICE_STATE() 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
  * A new option to Dial() for telling IP phones not to count the call
    as "missed" when dial times out and cancels.
  * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
    mutex.  No deadlocks are possible, as LOCK() only allows a single lock to be
    held for any given channel.  Also, locks are automatically freed when a
    channel is hung up.
  * Added HINT() dialplan function that allows retrieving hint information.
    Hints are mappings between extensions and devices for the sake of 
    determining the state of an extension.  This function can retrieve the list
    of devices or the name associated with a hint.
  * Added EXTENSION_STATE() dialplan function which allows retrieving the state
    of any extension.

CLI Changes
-----------
  * New CLI command "core show settings"
  * Added 'core show channels count' CLI command.
  * Added the ability to set the core debug and verbose values on a per-file basis.
  * Added 'queue pause member' and 'queue unpause member' CLI commands

SIP changes
-----------
  * Improved NAT and STUN support.
     chan_sip now can use port numbers in bindaddr, externip and externhost
     options, as well as contact a STUN server to detect its external address
     for the SIP socket. See sip.conf.sample, 'NAT' section.
  * 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.
  * Added rtpdest option to CHANNEL() dialplan function.
  * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
  * SIP now adds a header to the CANCEL if the call was answered by another phone
     in the same dial command, or if the new c option in dial() is used.
  * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
     states it is not needed. For phones, however, that do require it the registertrying option
     has been added so it can be enabled. 

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.

Skinny changes
-------------
  * Added skinny show device, skinny show line, and skinny show settings CLI commands.

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.

ENUM changes
------------
  * Added two new dialplan functions, ENUMQUERY and ENUMRESULT.  These
     functions will allow you to initiate an ENUM lookup from the dialplan,
     and Asterisk will cache the results.  ENUMRESULT can be used to access
     the results without doing multiple DNS queries.

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.
  * Added "tw" language support
  * Added support for storage of greetings using an IMAP server
  * Added ability to customize forward, reverse, stop, and pause keys for message playback
  * SMDI is now enabled in voicemail using the smdienable option.
  * A "lockmode" option has been added to asterisk.conf to configure the file
     locking method used for voicemail, and potentially other things in the
     future.  The default is the old behavior, lockfile.  However, there is a
     new method, "flock", that uses a different method for situations where the
     lockfile will not work, such as on SMB/CIFS mounts.

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.
  * Added ability for non-realtime queues to have realtime members
  * Added the "linear" strategy to queues.
  * Added the "wrandom" strategy to queues.

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.
  * Added 'C' option to Meetme which causes a caller to continue in the dialplan
     when kicked out.
  * Added new RealTime functionality to provide support for scheduled conferencing.
     This includes optional messages to the caller if they attempt to join before
     the schedule start time, or to allow the caller to join the conference early.
     Also included is optional support for limiting the number of callers per
     RealTime conference.
  * Added the S() and L() options to the MeetMe application.  These are pretty
     much identical to the S() and L() options to Dial().  They let you set
     timeouts for the conference, as well as have warning sounds played to
     let the caller know how much time is left, and when it is running out.

Music On Hold Changes
---------------------
  * A new option, "digit", has been added for music on hold classes in 
     musiconhold.conf.  If this is set for a music on hold class, a caller
     listening to music on hold can press this digit to switch to listening
     to this music on hold class.

AEL Changes
-----------
  * 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.
     -- in addition to this, all apps that allowed a macro
     to be called, as in Dial, queues, etc, are now allowing
     a gosub call in similar fashion.
  * AEL now generates LOCAL(argname) declarations when it
     Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
     etc. That makes the arguments local in scope. The user
     can define their own local variables in macros, now,
     by saying "local myvar=someval;"  or using Set() in this
     fashion:  Set(LOCAL(myvar)=someval);  ("local" is now
     an AEL keyword).
  * utils/conf2ael introduced. Will convert an extensions.conf
    file into extensions.ael. Very crude and unfinished, but 
    will be improved as time goes by. Should be useful for a
    first pass at conversion.
  * aelparse will now read extensions.conf to see if a referenced
    macro or context is there before issueing a warning.

Zaptel channel driver (chan_zap) Changes
----------------------------------------
  * SS7 support in chan_zap (via libss7 library)
  * In India, some carriers transmit CID via dtmf. Some code has been added
    that will handle some situations. The cidstart=polarity_IN choice has been added for
    those carriers that transmit CID via dtmf after a polarity change.
  * CID matching information is now shown when doing 'dialplan show'.
  * Added zap show version CLI command to chan_zap.
  * Added setvar support to zapata.conf channel entries.

H.323 Changes
-------------
  * H323 remote hold notification support added (by NOTIFY message
     and/or H.450 supplementary service)

Call Features (res_features) Changes
------------------------------------
  * Added the parkedcalltransfers option to features.conf
  * 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.
  * Added support for configuring named groups of custom call features in
     features.conf.  This means that features can be written a single time, and
     then mapped into groups of features for different key mappings or easier
     access control.

Language Support Changes
------------------------
  * Brazilian Portuguese (pt-BR) in VM, and say.c was added
  * Added support for the Hungarian language for saying numbers, dates, and times.

Miscellaneous 
-------------
  * Added the bindaddr option to gtalk.conf.
  * Argument support for Gosub application
  * Ability to set process limits without restarting Asterisk
  * Proper codec support in chan_skinny.
  * 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.
  * 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 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 's' option to Page application.
  * Added 'E' and 'V' commands to ExternalIVR.
  * Added 'o' and 'X' options to Chanspy.
  * Added a new CDR module, cdr_sqlite3_custom.
  * The cdr_manager module has a [mappings] feature, like cdr_custom,
    to add fields to the manager event from the CDR variables.
  * 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 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.
  * When originating a call using AMI or pbx_spool that fails the reason for failure
     will now be available in the failed extension using the REASON dialplan variable.
  * Added jitterbuffer support for chan_local.  This allows you to use the
     generic jitterbuffer on incoming calls going to Asterisk applications.
     For example, this would allow you to use a jitterbuffer for an incoming
     SIP call to Voicemail by putting a Local channel in the middle.  This
     feature is enabled by using the 'j' option in the Dial string to the Local
     channel in conjunction with the existing 'n' option for local channels.
  * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
     It allows you to configure a prefix for auto-monitor recordings.
  * Added support for writing and running your dialplan in lua.  See
     configs/extensions.lua.sample for examples of how to do this.
 * Added a new channel driver, chan_unistim.  See doc/unistim.txt and
    configs/unistim.conf.sample for details.  This new channel driver allows
    you to use Nortel i2002, i2004, and i2050 phones with Asterisk.