CHANGES

    realtime table lacks critical fields, Asterisk will now emit warnings to
    that effect.  Also, some of the realtime drivers have the ability (if
    configured) to automatically add those columns to the table with the
    correct type and length.

Miscellaneous
-------------
  * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
    the 'setvar' option to cause a given audio file to be played upon completion
    of an attended transfer.  Currently it works for DAHDI, IAX2, SIP, and
    Skinny channels only.
  * You can now compile Asterisk against the Hoard Memory Allocator, see the
    Hoard page on the Asterisk wiki for more information:
    https://wiki.asterisk.org/wiki/x/pQBB
  * Config file variables may now be appended to, by using the '+=' append
    operator.  This is most helpful when working with long SQL queries in
    func_odbc.conf, as the queries no longer need to be specified on a single
    line.
  * CDR config file, cdr.conf, has an added option, "initiatedseconds",
    which will add a second to the billsec when the ending
    time is set, if the number in the microseconds field of the end time is
    greater than the number of microseconds in the answer time. This allows
    users to count the 'initiated' seconds in their billing records.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0  -------------
------------------------------------------------------------------------------

AMI - The manager (TCP/TLS/HTTP)
--------------------------------
  * Manager has undergone a lot of changes, all of them documented
    on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
  * Manager version has changed to 1.1
  * Added a new action 'CoreShowChannels' to list currently defined channels
     and some information about them.
  * Added a new action 'SIPshowregistry' to list SIP registrations.
  * 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.
  * cdr_manager now reports events via the "cdr" level, separating it from
     the very verbose "call" level.
  * Manager users are now stored in memory. If you change the manager account
    list (delete or add accounts) you need to reload manager.
  * Added Masquerade manager event for when a masquerade happens between
     two channels.
  * Added "manager reload" command for the CLI
  * Lots of commands that only provided information are now allowed under the
     Reporting privilege, instead of only under Call or System.
  * The IAX* commands now require either System or Reporting privilege, to
     mirror the privileges of the SIP* commands.
  * Added ability to retrieve list of categories in a config file.
  * Added ability to retrieve the content of a particular category.
  * Added ability to empty a context.
  * Created new action to create a new file.
  * Updated delete action to allow deletion by line number with respect to category.
  * Added new action insert to add new variable to category at specified line.
  * Updated action newcat to allow new category to be inserted in file above another
    existing category.
  * Added new event "JitterBufStats" in the IAX2 channel
  * Originate now requires the Originate privilege and, if you want to call out
    to a subshell, it requires the System privilege, as well.  This was done to
    enhance manager security.
  * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
  * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
    or manager show command Atxfer from the CLI
  * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
    details or manager show command IAXregistry from the CLI

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.
  * Added SYSINFO() dialplan function which allows retrieval of system information
  * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
     the existence of a dialplan target.
  * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
     upper and lower case, respectively.
  * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
     ID for the call (not the Asterisk call ID or unique ID), provided that the
     channel driver supports this. For SIP, you get the SIP call-ID for the
     bridged channel which you can store in the CDR with a custom field.

CLI Changes
-----------
  * Added CLI permissions, config file: cli_permissions.conf
     default is to allow all commands for every local user/group.
     Also this new feature added three new CLI commands:
      - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
      - cli reload permissions
      - cli show permissions
  * New CLI command "core show hint" (usage: core show hint <exten>)
  * 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
  * Ability to set process limits ("ulimit") without restarting Asterisk
  * Enhanced "agi debug" to print the channel name as a prefix to the debug
     output to make debugging on busy systems much easier.
  * New CLI commands "dialplan set extenpatternmatching true/false"
  * New CLI command: "core set chanvar" to set a channel variable from the CLI.
  * Added an easy way to execute Asterisk CLI commands at startup.  Any commands
    listed in the startup_commands section of cli.conf will get executed.
  * Added a CLI command, "devstate change", which allows you to set custom device
     states from the func_devstate module that provides the DEVICE_STATE() function
     and handling of the "Custom:" devices.
  * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
    sorted into the different possible callbacks, with the number of entries
    currently scheduled for each. Gives you a feel for how busy the sip channel
    driver is.
  * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
  * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
    (Done by lmadsen, junky and mvanbaak during the devcon 2008)

SIP changes
-----------
 * Added a new 'faxdetect=yes|no' configuration option to sip.conf.  When this
    option is enabled, Asterisk will watch for a CNG tone in the incoming audio
    for a received call.  If it is detected, the channel will jump to the
    'fax' extension in the dialplan.
  * 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 "busylevel" for setting a level of calls where asterisk reports
     a device as busy, to separate it from call-limit. This value is also added
     to the SIP_PEER dialplan function.
  * 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.
  * A new option called "callcounter" (global/peer/user level) enables call counters needed
     for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
     used to enable this functionality).
  * New settings for timer T1 and timer B on a global level or per device. This makes it
     possible to force timeout faster on non-responsive SIP servers. These settings are
     considered advanced, so don't use them unless you have a problem.
  * Added a dial string option to be able to set the To: header in an INVITE to any
     SIP uri.
  * Added a new global and per-peer option, qualifyfreq, which allows you to configure
     the qualify frequency.
  * Added SIP Session Timers support (RFC 4028).  This prevents stuck SIP sessions that
     were not properly torn down due to network or endpoint failures during an established
     SIP session.
  * Added experimental TCP and TLS support for SIP.  See https://wiki.asterisk.org/wiki/x/ygBB
     and configs/sip.conf.sample for more information on how it is used.
  * Added a new configuration option "authfailureevents" that enables manager events when
    a peer can't authenticate properly.
  * Added DNS manager support to registrations for peers not referencing a peer entry.

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.

XMPP Google Talk/Jingle changes
-------------------------------
  * Added the bindaddr option to gtalk.conf.

Skinny changes
-------------
  * Added skinny show device, skinny show line, and skinny show settings CLI commands.
  * Proper codec support in chan_skinny.
  * Added settings for IP and Ethernet QoS requests

MGCP changes
------------
  * Added separate settings for media QoS in mgcp.conf

Console Channel Driver changes
------------------------------
  * Added experimental support for video send & receive to chan_oss.
    This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
    a video source.

Phone channel changes (chan_phone)
----------------------------------
  * Added G729 passthrough support to chan_phone for Sigma Designs boards.

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

Local channel changes
---------------------
  * 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 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.
  * A 'b' option has been added which causes chan_local to return the actual channel
     that is behind it when queried. This is useful for transfer scenarios as the
     actual channel will be transferred, not the Local channel.

Agent channel changes
----------------------
  * The ackcall and endcall options are now supplemented with options acceptdtmf
    and enddtmf. These allow for the DTMF keypress to be configurable. The options
    default to their old hard-coded values ('#' and '*' respectively) so this should
    not break any existing agent installations.

DAHDI channel driver (chan_dahdi) Changes
----------------------------------------
  * SS7 support (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 dahdi show version CLI command.
  * Added setvar support to chan_dahdi.conf channel entries.
  * Added two new options: mwimonitor and mwimonitornotify.  These options allow
     you to enable MWI monitoring on FXO lines.  When the MWI state changes,
     the script specified in the mwimonitornotify option is executed.  An internal
     event indicating the new state of the mailbox is also generated, so that
     the normal MWI facilities in Asterisk work as usual.
  * Added signalling type 'auto', which attempts to use the same signalling type
     for a channel as configured in DAHDI. This is primarily designed for analog
     ports, but will also work for digital ports that are configured for FXS or FXO
     signalling types. This mode is also the default now, so if your chan_dahdi.conf
     does not specify signalling for a channel (which is unlikely as the sample
     configuration file has always recommended specifying it for every channel) then
     the 'auto' mode will be used for that channel if possible.
  * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
     state for a channel; also ensured that the DNDState Manager event is
     emitted no matter how the DND state is set or cleared.

New Channel Drivers
-------------------
  * Added a new channel driver, chan_unistim.  See the Asterisk wiki at
     https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
     for details.  This new channel driver allows you to use Nortel i2002,
     i2004, and i2050 phones with Asterisk.
  * Added a new channel driver, chan_console, which uses portaudio as a cross
     platform audio interface.  It was written as a channel driver that would
     work with Mac CoreAudio, but portaudio supports a number of other audio
     interfaces, as well. Note that this channel driver requires v19 or higher
     of portaudio; older versions have a different API.

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.
  * Added the ability to specifiy a port for a dundi peer.

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.
  * Added the ability to backup deleted messages, to ease recovery in the case
     that a user accidentally deletes a message, and discovers that they need it.
  * Reworked the SMDI interface in Asterisk.  The new way to access SMDI information
     is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG().  The file
     smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
     voicemail boxes.  The SMDI interface can also poll for MWI changes when some
     outside entity is modifying the state of the mailbox (such as IMAP storage or
     a web interface of some kind).
  * Added the support for marking messages as "urgent." There are two methods to accomplish
     this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
     is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
     the message as urgent after he has recorded a voicemail by following the voice instructions.
    When listening to voicemails using VoiceMailMain urgent messages will be presented before other
     messages
  * Added "is" language support

Queue changes
-------------
  * Added the general option 'shared_lastcall' so that member's wrapuptime may be
     used across multiple queues.
  * 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.
  * Added new channel variable QUEUE_MIN_PENALTY
  * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
     rules in queuerules.conf. See configs/queuerules.conf.sample for details
  * Added a new parameter for member definition, called state_interface. This may be
    used so that a member may be called via one interface but have a different interface's
    device state reported.
  * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
    "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
    "manager show command QueueReset."
  * New configuration option: randomperiodicannounce. If a list of periodic announcements is
    specified by the periodic-announce option, then one will be chosen randomly when it is time
    to play a periodic announcment
  * New configuration options: announce-position now takes two more values in addition to "yes" and
    "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
    announce-position-limit. By setting announce-position to "limit" callers will only have their
    position announced if their position is less than what is specified by announce-position-limit.
    If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
    will be told that their are more than announce-position-limit callers waiting.
  * Two new queue log events have been added. An ADDMEMBER event will be logged
    when a realtime queue member is added and a REMOVEMEMBER event will be logged
    when a realtime queue member is removed. Since there is no calling channel associated
    with these events, the string "REALTIME" is placed where the channel's unique id
    is typically placed.
  * The configuration method for the "joinempty" and "leavewhenempty" options has
    changed to a comma-separated list of methods of determining member availability
    instead of vague terms such as "yes," "loose," "no," and "strict." These old four
    values are still accepted for backwards-compatibility, though.
  * The average talktime is now calculated on queues. This information is reported via the
    CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
    and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
    the queue.

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.
  * Added the ability to do "meetme concise" with the "meetme" CLI command.
     This extends the concise capabilities of this CLI command to include
     listing all conferences, instead of an addition to the other sub commands
     for the "meetme" command.
  * Added the ability to specify the music on hold class used to play into the
     conference when there is only one member and the M option is used.
  * Added MEETME_INFO dialplan function which provides a way to query
     various properties of a Meetme conference.
  * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
     and *84: record in-conf

Other Dialplan Application Changes
----------------------------------
  * Argument support for Gosub application
  * 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 's' option to Page application.
  * Added an optional timeout argument to the Page application.
  * Added 'E', 'V', and 'P' commands to ExternalIVR.
  * Added 'o' and 'X' options to Chanspy.
  * Added a new dialplan application, Bridge, which allows you to bridge the
     calling channel to any other active channel on the system.
  * Added the ability to specify a music on hold class to play instead of ringing
     for the SLATrunk application.
  * The Read application no longer exits the dialplan on error.  Instead, it sets
     READSTATUS to ERROR, which you can catch and handle separately.
  * Added 'm' option to Directory, which lists out names, 8 at a time, instead
     of asking for verification of each name, one at a time.
  * Privacy() no longer uses privacy.conf, as all options are specifyable as
     direct options to the app.
  * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
     for more details
  * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
  * The ChannelRedirect application no longer exits the dialplan if the given channel
     does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
     or NOCHANNEL if the given channel was not found.
  * The silencethreshold setting that was previously configurable in multiple
     applications is now settable globally via dsp.conf.

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.
  * Support for realtime music on hold has been added.
  * In conjunction with the realtime music on hold, a general section has
     been added to musiconhold.conf, its sole variable is cachertclasses. If this
     is set, then music on hold classes found in realtime will be cached in memory.

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.
  * AEL parser sets a local channel variable ~~EXTEN~~, to
    preserve the value of ${EXTEN} thru switch statements.
  * New operator in $[...] expressions: the ~~ operator serves
    as a concatenation operator. AT THE MOMENT, it is really only
    necessary and useful in AEL, especially in if() expressions.
    Operation: ${a} ~~ ${b|  with force both a and b to strings, strip
    any enclosing double-quotes, and evaluate to the value of a
    concatenated with the value of b.  For example if a is set to
    "xyz"  and b has the value "abc", then ${a} ~~ ${b| would
    evaluate to xyzabc .


Call Features (res_features) Changes
------------------------------------
  * Added the parkedcalltransfers option to features.conf
  * Added parkedcallparking option to control one touch parking w/ parking
    pickup
  * Added parkedcallhangup option to control disconnect feature w/ parking
    pickup
  * Added parkedcallrecording option to control one-touch record w/ parking
    pickup
  * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
    parkedcalltransfers option support for multiple parking lots.
  * Added BRIDGE_FEATURES variable to set available features for a channel
  * 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.
  * Updated the ParkedCall application to allow you to not specify a parking
     extension.  If you don't specify a parking space to pick up, it will grab
     the first one available.
  * Added cli command 'features reload' to reload call features from features.conf
  * Moved into core asterisk binary.
  * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
  * Added the ability for custom parking lots to be configured with their own
    parking extension with the parkext option.

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.

AGI Changes
-----------
  * Added SPEECH commands for speech recognition. A complete listing can be found
    using agi show.
  * If app_stack is loaded, GOSUB is a native AGI command that may be used to
    invoke subroutines in the dialplan.  Note that calling EXEC with Gosub
    does not behave as expected; the native command needs to be used, instead.
  * Added the ability to perform SRV lookups on fast AGI calls. To use this
    feature, simply use hagi: instead of agi: as the protocol portion
    of the URI parameter to the AGI function call in your dial plan. Also note
    that specifying a port number in the AGI URI will disable SRV lookups,
    even if you use the hagi: protocol.
  * No longer support MSG_OOB flag on HANGUP.

Logger changes
--------------
  * Added rotatestrategy option to logger.conf, along with two new options:
     "timestamp" which will use the time to name the logger files instead of
     sequence number; and "rotate", which rotates the names of the log files,
     similar to the way syslog rotates files.
  * Added exec_after_rotate option to logger.conf, which allows a system
     command to be run after rotation.  This is primarily useful with
     rotatestrategy=rotate, to allow a limit on the number of log files kept
     and to ensure that the oldest log file gets deleted.
  * Added realtime support for the queue log

Call Detail Records
-------------------
  * The cdr_manager module has a [mappings] feature, like cdr_custom,
    to add fields to the manager event from the CDR variables.
  * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
     backend database CDR table.  Specifically, additional, non-standard
     columns are supported, merely by setting the corresponding CDR variable in
     your dialplan.  In addition, you may alias any column to another name (for
     example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
     simply "alias src => ANI" in the configuration file).  Records may be
     posted to more than one backend, simply by specifying multiple categories
     in the configuration file.  And finally, you may filter which CDRs get
     posted to each backend, by specifying a filter (which the record must
     match) for the particular category.  Filters are additive (meaning all
     rules must match to post that CDR).
  * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
     module.  Specifically, you may add additional columns into the table and
     they will be set, if you set the corresponding CDR variable name.  Also,
     if you omit columns in your database table, they will be silently skipped
     (but a record will still be inserted, based on what columns remain).  Note
     that the other two features from cdr_adaptive_odbc (alias and filter) are
     not currently supported.
  * The ResetCDR application now has an 'e' option that re-enables a CDR if it
     has been disabled using the NoCDR application.

Miscellaneous New Modules
-------------------------
  * Added a new CDR module, cdr_sqlite3_custom.
  * Added a new realtime configuration module, res_config_sqlite
  * Added a new codec translation module, codec_resample, which re-samples
     signed linear audio between 8 kHz and 16 kHz to help support wideband
     codecs.
  * Added a new module, res_phoneprov, which allows auto-provisioning of phones
     based on configuration templates that use Asterisk dialplan function and
     variable substitution.  It should be possible to create phone profiles and
     templates that work for the majority of phones provisioned over http. It
     is currently only intended to provision a single user account per phone.
     An example profile and set of templates for Polycom phones is provided.
     NOTE: Polycom firmware is not included, but should be placed in
     AST_DATA_DIR/phoneprov/configs to match up with the included templates.
  * Added a new module, app_jack, which provides interfaces to JACK, the Jack
     Audio Connection Kit (http://www.jackaudio.org/).  Two interfaces are
     provided; there is a JACK() application, and a JACK_HOOK() function.  Both
     interfaces create an input and output JACK port.  The application makes
     these ports the endpoint of the call.  The audio coming from the channel
     goes out the output port and whatever comes back in on the input port is
     what gets sent to the channel.  The JACK_HOOK() function turns on a JACK
     audiohook on the channel.  This lets you run the audio coming from a
     channel through JACK, and whatever comes back in is what gets forwarded
     on as the channel's audio.  This is very useful for building custom
     vocoders or doing recording or analysis of the channel's audio in another
     application.
  * Added a new module, res_config_curl, which permits using a HTTP POST url
     to retrieve, create, update, and delete realtime information from a remote
     web server.  Note that this module requires func_curl.so to be loaded for
     backend functionality.
  * Added a new module, res_config_ldap, which permits the use of an LDAP
     server for realtime data access.
  * Added support for writing and running your dialplan in lua using the pbx_lua
     module.  See configs/extensions.lua.sample for examples of how to do this.

Miscellaneous
-------------
  * 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 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 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 on the Asterisk wiki at
     https://wiki.asterisk.org/wiki/x/EYBG
  * 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 support for reading the TOUCH_MONITOR_PREFIX channel variable.
     It allows you to configure a prefix for auto-monitor recordings.
  * A new extension pattern matching algorithm, based on a trie, is introduced
     here, that could noticeably speed up mid-sized to large dialplans.
     It is NOT used by default, as duplicating the behaviour of the old pattern
     matcher is still under development. A config file option, in extensions.conf,
     in the [general] section, called "extenpatternmatchingnew", is by default
     set to false; setting that to true will force the use of the new algorithm.
     Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
     be used to switch the algorithms at run time.
  * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
     specifying which socket to use to connect to the running Asterisk daemon
     (-s)
  * Performance enhancements to the sched facility, which is used in
    the channel drivers, etc. Added hashtabs and doubly-linked lists
    to speed up deletion; start at the beginning or end of list to
    speed up insertion.
  * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
    dlinkedlists.h. Doubly-linked lists feature fast deletion times.
    Added regression tests to the tests/ dir, also.
  * Added a refcount trace feature to astobj2 for those trying to balance
    object creation, deletion; work, play; space and time. See the
    notes in astobj2.h. Also, see utils/refcounter as well, as a
    quick way to find unbalanced refcounts in what could be a sea
    of objects that were balanced.
  * Added logging to 'make update' command.  See update.log
  * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
     do not come from the remote party.
  * Added the 'n' option to the SpeechBackground application to tell it to not
     answer the channel if it has not already been answered.
  * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
     turned on, via the CHANNEL(trace) dialplan function.  Could be useful for
     dialplan debugging.
  * iLBC source code no longer included (see UPGRADE.txt for details)
  * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
     deadlock is detected, a backtrace of the stack which led to the lock calls
     will be output to the CLI.
  * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
     the "core show locks" CLI command will give lock information output as well
     as a backtrace of the stack which led to the lock calls.
  * users.conf now sports an optional alternateexts property, which permits
    allocation of additional extensions which will reach the specified user.
  * A new option for the configure script, --enable-internal-poll, has been added
    for use with systems which may have a buggy implementation of the poll system
    call. If you notice odd behavior such as the CLI being unresponsive on remote
    consoles, you may want to try using this option. This option is enabled by default
    on Darwin systems since it is known that the Darwin poll() implementation has
    odd issues.

Timer Changes
--------------------
* In addition to timing from DAHDI, there is a new timing module called
  res_timing_timerfd. In order to use this, you must be running Linux with
  a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
  script will be able to tell if you have the requirements. From menuselect, select
  res_timing_timerfd from the Resource Modules menu.