CHANGES

==============================================================================
===
=== This file documents the new and/or enhanced functionality added in
=== the Asterisk versions listed below. This file does NOT include
=== changes in behavior that would not be backwards compatible with
=== previous versions; for that information see the UPGRADE.txt file
=== and the other UPGRADE files for older releases.
===
==============================================================================

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
------------------------------------------------------------------------------

Build System
-------------------
 * The Asterisk build system will now build and install a shared library
   (libasteriskssl.so) used to wrap various initialization and shutdown functions
   from the libssl and libcrypto libraries provided by OpenSSL. This is done so
   that Asterisk can ensure that these functions do *not* get called by any
   modules that are loaded into Asterisk, since they should only be called once
   in any single process. If desired, this feature can be disabled by supplying
   the "--disable-asteriskssl" option to the configure script.

 * A new make target, 'full', has been added to the Makefile.  This performs
   the same compilation actions as make all, but will also scan the entirety of
   each source file for documentation.  This option is needed to generate AMI
   event documentation.  Note that your system must have Python in order for
   this make target to succeed.

 * The optimization portion of the build system has been reworked to avoid
   broken builds on certain architectures.  All architecture-specific
   optimization has been removed in favor of using -march=native to allow gcc
   to detect the environment in which it is running when possible.  This can
   be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.

 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
   make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"

 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h".  If you
   previously parsed the header file to obtain the version of Asterisk, you
   will now have to go through Asterisk to get the version information.


Applications
-------------------

Bridge
-------------------
 * Added 'F()' option. Similar to the dial option, this can be supplied with
   arguments indicating where the callee should go after the caller is hung up,
   or without options specified, the priority after the Queue will be used.


ConfBridge
-------------------
 * Added menu action admin_toggle_mute_participants.  This will mute / unmute
   all non-admin participants on a conference.  The confbridge configuration
   file also allows for the default sounds played to all conference users when
   this occurs to be overriden using sound_participants_unmuted and
   sound_participants_muted.

 * Added menu action participant_count.  This will playback the number of
   current participants in a conference.

 * Added announcement configuration option to user profile. If set the sound
   file will be played to the user, and only the user, upon joining the
   conference bridge.


Dial
-------------------
 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
   channels respectively before the callee channels are called.


ExternalIVR
-------------------
 * Added support for IPv6.

 * Add interrupt ('I') command to ExternalIVR.  Sending this command from an 
   external process will cause the current playlist to be cleared, including
   stopping any audio file that is currently playing.  This is useful when you
   want to interrupt audio playback only when specific DTMF is entered by the
   caller.


FollowMe
-------------------
 * A new option, 'I' has been added to app_followme. By setting this option,
   Asterisk will not update the caller with connected line changes when they
   occur.  This is similar to app_dial and app_queue.

 * The 'N' option is now ignored if the call is already answered.

 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
   and caller channels respectively before the callee channels are called.

 * The winning FollowMe outgoing call is now put on hold if the caller put it on
   hold.


MixMonitor
------------------
 * MixMonitor hooks now have IDs associated with them which can be used to
   assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
   will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
   now accepts that ID as an argument.

 * Added 'm' option, which stores a copy of the recording as a voicemail in the
   indicated mailboxes.


OSP Applications
-------------------
 * Increased the default number of allowed destinations from 5 to 12.


Page
-------------------
 * The app_page application now no longer depends on DAHDI or app_meetme.  It
   has been re-architected to use app_confbridge internally.


Queue
-------------------
 * Added queue options autopausebusy and autopauseunavail for automatically
   pausing a queue member when their device reports busy or congestion.

 * The 'ignorebusy' option for queue members has been deprecated in favor of
   the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
   added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
   per interface basis. Individual ringinuse values can now be set in
   queues.conf via an argument to member definitions. Lastly, the queue
   'ringinuse' setting now only determines defaults for the per member
   'ringinuse' setting and does not override per member settings like it does
   in earlier versions.

 * Added 'F()' option. Similar to the dial option, this can be supplied with
   arguments indicating where the callee should go after the caller is hung up,
   or without options specified, the priority after the Queue will be used.

 * Added new option log_member_name_as_agent, which will cause the membername to
   be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
   state_interface has been set.


SayUnixTime
------------------
 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
   when receiving DTMF.  Use the 'j' option to enable extension jumping. Also
   changed arguments to SayUnixTime so that every option is truly optional even
   when using multiple options (so that j option could be used without having to
   manually specify timezone and format) There are other benefits, e.g., format
   can now be used without specifying time zone as well.


Voicemail
------------------
 * Addition of the VM_INFO function - see Function changes.

 * The imapserver, imapport, and imapflags configuration options can now be
   overriden on a user by user basis.

 * When voicemail plays a message's envelope with saycid set to yes, when
   reaching the caller id field it will play a recording of a file with the same
   base name as the sender's callerid if there is a similarly named file in
   <astspooldir>/recordings/callerids/

 * Voicemails now contains a unique message identifier "msg_id", which is stored
   in the message envelope with the sound files.  IMAP backends will now store
   the message identifiers with a header of "X-Asterisk-VM-Message-ID".  ODBC
   backends will store the message identifier in a "msg_id" column.  See
   UPGRADE.txt for more information.

 * Added VoiceMailPlayMsg application.  This application will play a single
   voicemail message from a mailbox.  The result of the application, SUCCESS or
   FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.


Functions
------------------
 * Hangup handlers can be attached to channels using the CHANNEL() function.
   Hangup handlers will run when the channel is hung up similar to the h
   extension. The hangup_handler_push option will push a GoSub compatible
   location in the dialplan onto the channel's hangup handler stack.  The
   hangup_handler_pop option will remove the last added location, and optionally
   replace it with a new GoSub compatible location.  The hangup_handler_wipe
   option will remove all locations on the stack, and optionally add a new
   location.

 * The expression parser now recognizes the ABS() absolute value function,
   which will convert negative floating point values to positive values.

 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
   control of faxdetect.

 * Addition of the VM_INFO function that can be used to retrieve voicemail
   user information, such as the email address and full name.
   The MAILBOX_EXISTS dialplan function has been deprecated in favour of
   VM_INFO.

 * The REDIRECTING function now supports the redirecting original party id
   and reason.

 * Two new functions have been added: FEATURE() and FEATUREMAP().  FEATURE()
   lets you set some of the configuration options from the [general] section
   of features.conf on a per-channel basis.  FEATUREMAP() lets you customize
   the key sequence used to activate built-in features, such as blindxfer,
   and automon.  See the built-in documentation for details.

 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
   instead of simply the uri.  This is the format that MessageSend() can use
   in the from parameter for outgoing SIP messages.

 * Added the PRESENCE_STATE function.  This allows retrieving presence state
   information from any presence state provider.  It also allows setting
   presence state information from a CustomPresence presence state provider.
   See AMI/CLI changes for related commands.


Channel Drivers
------------------

chan_local
------------------
 * Added a manager event "LocalBridge" for local channel call bridges between
   the two pseudo-channels created.


chan_dahdi
------------------
 * Added dialtone_detect option for analog ports to disconnect incoming
   calls when dialtone is detected.

 * Added option colp_send to send ISDN connected line information.  Allowed
   settings are block, to not send any connected line information; connect, to
   send connected line information on initial connect; and update, to send
   information on any update during a call.  Default is update.


chan_motif
------------------
 * A new channel driver named chan_motif has been added which provides support for
   Google Talk and Jingle in a single channel driver. This new channel driver includes
   support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
   hold, unhold, and ringing notification. It is also compliant with the current Jingle
   specification, current Google Jingle specification, and the original Google Talk
   protocol.


chan_ooh323
------------------
 * Added NAT support for RTP.  Setting in config is 'nat', which can be set
   globally and overriden on a peer by peer basis.

 * Direct media functionality has been added. Options in config are: 
   directmedia (directrtp) and directrtpsetup (earlydirect)

 * ChannelUpdate events now contain a CallRef header.


chan_sip
------------------
 * Asterisk will no longer substitute CID number for CID name in the display
   name field if CID number exists without a CID name. This change improves
   compatibility with certain device features such as Avaya IP500's directory
   lookup service.

 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
   created using that setting to not be removed during SIP reload.

 * Added settings recordonfeature and recordofffeature.  When receiving an INFO
   request with a "Record:" header, this will turn the requested feature on/off.
   Allowed values are 'automon', 'automixmon', and blank to disable.  Note that
   dynamic features must be enabled and configured properly on the requesting
   channel for this to function properly.

 * Add support to realtime for the 'callbackextension' option.

 * When multiple peers exist with the same address, but differing
   callbackextension options, incoming requests that are matched by address
   will be matched to the peer with the matching callbackextension if it is
   available.

 * Two new NAT options, auto_force_rport and auto_comedia, have been added
   which set the force_rport and comedia options automatically if Asterisk
   detects that an incoming SIP request crossed a NAT after being sent by
   the remote endpoint.

 * NAT settings are now a combinable list of options. The equivalent of the
   deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.

 * Adds an option send_diversion which can be disabled to prevent
   diversion headers from automatically being added to INVITE requests.

 * Add support for lightweight NAT keepalive. If enabled a blank packet will
   be sent to the remote host at a given interval to keep the NAT mapping open.
   This can be enabled using the keepalive configuration option.

 * Add option 'tonezone' to specify country code for indications.  This option
   can be set both globally and overridden for specific peers.

 * The SIP Security Events Framework now supports IPv6.

 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
   between multiple user agents. When set, for directmedia reinvites,
   Asterisk will not send an immediate reinvite on an incoming call leg. This
   option is useful when peered with another SIP user agent that is known to
   send immediate direct media reinvites upon call establishment.

 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
   as the transport.

 * When a MESSAGE request is received, the address the request was received from
   is now saved in the SIP_RECVADDR variable.

 * Add ANI2/OLI parsing for SIP.  The "From" header in INVITE requests is now
   parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags.  If present,
   the ANI2/OLI information is set on the channel, which can be retrieved using
   the CALLERID function.

 * Peers can now be configured to support negotiation of ICE candidates using
   the setting icesupport.  See res_rtp_asterisk changes for more information.

 * Added support for format attribute negotiation.  See the Codecs changes for
   more information.


chan_skinny
------------------
 * Added skinny version 17 protocol support.


chan_unistim
--------------------
 * Added ability to use multiple lines for a single phone.  This allows multiple
   calls to occur on a single phone, using callwaiting and switching between calls.

 * Added option 'sharpdial' allowing end dialing by pressing # key

 * Added option 'interdigit_timer' to control phone dial timeout

 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance

 * Added global 'debug' option, that enables debug in channel driver

 * Added ability to translate on-screen menu in multiple languages. Tested on
   Russian languages.  Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4, 
   ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen 
   menu of phone

 * In addition to English added French and Russian languages for on-screen menus

 * Reworked dialing number input: added dialing by timeout, immediate dial on 
   on dialplan compare, phone number length now not limited by screen size

 * Added ability to pickup a call using features.conf defined value and 
   on-screen key


Core
------------------
 * The minimum DTMF duration can now be configured in asterisk.conf
   as "mindtmfduration". The default value is (as before) set to 80 ms.
   (previously it was only available in source code)

 * Named ACLs can now be specified in acl.conf and used in configurations that
   use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
   used to specify an ACL, a similar form of 'acl' will add a named ACL to the
   working ACL. In addition, some CLI commands have been added to provide
   show information and allow for module reloading - see CLI Changes.

 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
   be used within the dynamic weight attribute when specifying a mapping.

 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
   header, instead of putting the user defined event name there.  When enabled
   the UserDefType header is added for user defined events.  This feature is
   enabled with the setting show_user_defined.

 * Macro has been deprecated in favor of GoSub.  For redirecting and connected
   line purposes use the following variables instead of their macro equivalents:
   REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
   CONNECTED_LINE_SEND_SUB_ARGS.  For CCSS, use cc_callback_sub instead of
   cc_callback_macro in channel configurations.


AGI
------------------
 * A new channel variable, AGIEXITONHANGUP, has been added which allows
   Asterisk to behave like it did in Asterisk 1.4 and earlier where the
   AGI application would exit immediately after a channel hangup is detected.

 * IPv6 addresses are now supported when using FastAGI (agi://).  Hostnames
   are resolved and each address is attempted in turn until one succeeds or
   all fail.


AMI (Asterisk Manager Interface)
------------------
 * Originate now generates an error response if the extension given is not found
   in the dialplan

 * MixMonitor will now show IDs associated with the mixmonitor upon creating
   them if the i(variable) option is used. StopMixMonitor will accept
   MixMonitorID as an option to close specific MixMonitors.

 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
   updated to include information about peers configured with
   nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
   detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
   returned if auto_force_rport is not enabled.

 * Hangup now can take a regular expression as the Channel option.  If you want
   to hangup multiple channels, use /regex/ as the Channel option.  Existing 
   behavior to hanging up a single channel is unchanged, but if you pass a regex,
   the manager will send you a list of channels back that were hung up.

 * Support for IPv6 addresses has been added.

 * AMI Events can now be documented in the Asterisk source. Note that AMI event
   documentation is only generated when Asterisk is compiled using 'make full'.
   See the CLI section for commands to display AMI event information.

 * The AMI Hangup event now includes the AccountCode header so you can easily
   correlate with AMI Newchannel events.

 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
   the StateInterface of the queue member.

 * Added AMI event SessionTimeout in the Call category that is issued when a
   call is terminated due to either RTP stream inactivity or SIP session timer
   expiration.

 * CEL events can now contain a user defined header UserDefType.  See core
   changes for more information.

 * OOH323 ChannelUpdate events now contain a CallRef header.

 * Added PresenceState command.  This command will report the presence state for
   the given presence provider.

 * Added Parkinglots command.  This will list all parking lots as a series of
   AMI Parkinglot events.

 * Added MessageSend command.  This behaves in the same manner as the
   MessageSend application, and is a technolgoy agnostic mechanism to send out
   of call text messages.

 * Added "message" class authorization.  This grants an account permission to
   send out of call messages.  Write-only.


CLI
-------------------
 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
   filenames of all running mixmonitors on a channel.

 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
   numeric instead of 0, 1, or 2.

 * "stun show status" will show a table describing how the STUN client is
   behaving.

 * "acl show [named acl]" will show information regarding a Named ACL.  The
   acl module can be reloaded with "reload acl".

 * Added CLI command to display AMI event information - "manager show events",
   which shows a list of all known and documented AMI events, and "manager show
   event [event name]", which shows detail information about a specific AMI
   event.

 * The result of the CLI command "queue show" now includes the state interface
   information of the queue member.

 * The command "core set verbose" will now set a separate level of logging for
   each remote console without affecting any other console.

 * Added command "cdr show pgsql status" to check connection status

 * "sip show channel" will now display the complete route set.

 * Added "presencestate list" command.  This command will list all custom
   presence states that have been set by using the PRESENCE_STATE dialplan
   function.

 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
   command.  This changes a custom presence to a new state.


Codecs
-------------------
 * Codec lists may now be modified by the '!' character, to allow succinct
   specification of a list of codecs allowed and disallowed, without the
   requirement to use two different keywords.  For example, to specify all
   codecs except g729 and g723, one need only specify allow=all,!g729,!g723.

 * Add support for parsing SDP attributes, generating SDP attributes, and
   passing it through. This support includes codecs such as H.263, H.264, SILK,
   and CELT. You are able to set up a call and have attribute information pass.
   This should help considerably with video calls.


Logging
-------------------
 * Asterisk version and build information is now logged at the beginning of a
   log file.

 * Threads belonging to a particular call are now linked with callids which get
   added to any log messages produced by those threads. Log messages can now be
   easily identified as involved with a certain call by looking at their call id.
   Call ids may also be attached to log messages for just about any case where
   it can be determined to be related to a particular call.

 * Each logging destination and console now have an independent notion of the
   current verbosity level.  Logger.conf now allows an optional argument to
   the 'verbose' specifier, indicating the level of verbosity sent to that
   particular logging destination.  Additionally, remote consoles now each
   have their own verbosity level.  The command 'core set verbose' will now set
   a separate level for each remote console without affecting any other
   console.


Music On Hold
-------------------
 * Added 'announcement' option which will play at the start of MOH and between
   songs in modes of MOH that can detect transitions between songs (eg.
   files, mp3, etc).


Parking
-------------------
 * New per parking lot options: comebackcontext and comebackdialtime. See
   configs/features.conf.sample for more details.

 * Channel variable PARKER is now set when comebacktoorigin is disabled in
   a parking lot.

 * Channel variable PARKEDCALL is now set with the name of the parking lot 
   when a timeout occurs.


CDRs
-------------------

CDR Postgresql Driver
-------------------
 * Added command "cdr show pgsql status" to check connection status


CDR Adaptive ODBC Driver
-------------------
 * Added schema option for databases that support specifying a schema.


Resource Modules
-------------------

Calendars
-------------------
 * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not 
   CALENDAR_WRITE has completed successfully.


res_rtp_asterisk
-------------------
 * A new option, 'probation' has been added to rtp.conf
   RTP in strictrtp mode can now require more than 1 packet to exit learning
   mode with a new source (and by default requires 4). The probation option
   allows the user to change the required number of packets in sequence to any
   desired value. Use a value of 1 to essentially restore the old behavior.
   Also, with strictrtp on, Asterisk will now drop all packets until learning
   mode has successfully exited. These changes are based on how pjmedia handles
   media sources and source changes.

 * Add support for ICE/STUN/TURN in res_rtp_asterisk.  This option can be
   enabled or disabled using the icesupport setting.  A variety of other
   settings have been introduced to configure STUN/TURN connections.


res_corosync
-------------------
 * A new module, res_corosync, has been introduced.  This module uses the
   Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
   of Asterisk servers to both Message Waiting Indication (MWI) and/or
   Device State (presence) information.  This module is very similar to, and
   is a replacement for the res_ais module that was in previous releases of
   Asterisk.


res_xmpp
-------------------
 * This module adds a cleaned up, drop-in replacement for res_jabber called
   res_xmpp. This provides the same externally facing functionality but is
   implemented differently internally.  res_jabber has been deprecated in favor
   of res_xmpp; please see the UPGRADE.txt file for more information.


Scripts
-------------------
 * The safe_asterisk script has been updated to allow several of its parameters
   to be set from environment variables.  This also enables a custom run
   directory of Asterisk to be specified, instead of defaulting to /tmp.

 * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
   its value to determine the directory to assume is the top-level directory of
   the source tree.  If the variable is not set, it defaults to the current
   behavior and uses the current working directory.


------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
------------------------------------------------------------------------------

Text Messaging
--------------
 * Asterisk now has protocol independent support for processing text messages
   outside of a call.  Messages are routed through the Asterisk dialplan.
   SIP MESSAGE and XMPP are currently supported.  There are options in
   jabber.conf and sip.conf to allow enabling these features.
     -> jabber.conf: see the "sendtodialplan" and "context" options.
     -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
        and "outofcall_message_context" options.
   The MESSAGE() dialplan function and MessageSend() application have been
   added to go along with this functionality.  More detailed usage information
   can be found on the Asterisk wiki (http://wiki.asterisk.org/).
 * If real-time text support (T.140) is negotiated, it will be preferred for
   sending text via the SendText application. For example, via SIP, messages
   that were once sent via the SIP MESSAGE request would be sent via RTP if
   T.140 text is negotiated for a call.

Parking
-------
 * parkedmusicclass can now be set for non-default parking lots.

Asterisk Manager Interface
--------------------------
 * PeerStatus now includes Address and Port.
 * Added Hold events for when the remote party puts the call on and off hold
   for chan_dahdi ISDN channels.
 * Added new action MeetmeListRooms to list active conferences (shows same
   data as "meetme list" at the CLI).
 * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
   Description field that is set by 'description' in the channel configuration
   file.
 * Added Uniqueid header to UserEvent.
 * Added new action FilterAdd to control event filters for the current session.
   This requires the system permission and uses the same filter syntax as
   filters that can be defined in manager.conf
 * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
   versions had some instances of the event converted, but others were left
   as-is. All Unlink events should now be converted to Bridge events. The AMI
   protocol version number was incremented to 1.2 as a result of this change.

Asterisk HTTP Server
--------------------------
 * The HTTP Server can bind to IPv6 addresses.

chan_dahdi
--------------------------
 * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
   with busydetect.  usage example: busypattern=200,200,200,600

CLI Changes
--------------------------
 * New 'gtalk show settings' command showing the current settings loaded from
   gtalk.conf.
 * The 'logger reload' command now supports an optional argument, specifying an
   alternate configuration file to use.
 * 'dialplan add extension' command will now automatically create a context if
   the specified context does not exist with a message indicated it did so.
 * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
   Description field which can be populated with 'description' in the channel
   configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).

CDR
--------------------------
 * The filter option in cdr_adaptive_odbc now supports negating the argument,
   thus allowing records which do NOT match the specified filter.
 * Added ability to log CONGESTION calls to CDR

CODECS
--------------------------
 * Ability to define custom SILK formats in codecs.conf.
 * Addition of speex32 audio format with translation.
 * CELT codec pass-through support and ability to define
   custom CELT formats in codecs.conf.
 * Ability to read raw signed linear files with sample rates
   ranging from 8khz - 192khz.  The new file extensions introduced
   are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
 * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
   Skinny, H.323, etc) can still only support the following codecs:
   Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
          siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
   Video: h261, h263, h263p, h264, mpeg4
   Image: jpeg, png
   Text:  red, t140

ConfBridge
--------------------------
 * New highly optimized and customizable ConfBridge application capable of
   mixing audio at sample rates ranging from 8khz-96khz.
 * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
   and bridge profiles on a channel.
 * CONFBRIDGE_INFO dialplan function capable of retrieving information 
   about a conference such as locked status and number of parties, admins,
   and marked users.
 * Addition of video_mode option in confbridge.conf for adding video support
   into a bridge profile.
 * Addition of the follow_talker video_mode in confbridge.conf.  This video
   mode dynamically switches the video feed to always display the loudest talker
   supplying video in the conference.

Dialplan Variables
------------------
 * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
   ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
   variables from asterisk.conf.

Dialplan Functions
------------------
 * Addition of the JITTERBUFFER dialplan function. This function allows
   for jitterbuffering to occur on the read side of a channel.  By using
   this function conference applications such as ConfBridge and MeetMe can
   have the rx streams jitterbuffered before conference mixing occurs.
 * Added DB_KEYS, which lists the next set of keys in the Asterisk database
   hierarchy.
 * Added STRREPLACE function.  This function let's the user search a variable
   for a given string to replace with another string as many times as the
   user specifies or just throughout the whole string.
 * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
 * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
 * Added extensions to chan_ooh323 in function CHANNEL()

libpri channel driver (chan_dahdi) DAHDI changes
--------------------------
 * Added moh_signaling option to specify what to do when the channel's bridged
   peer puts the ISDN channel on hold.
 * Added display_send and display_receive options to control how the display ie
   is handled.  To send display text from the dialplan use the SendText()
   application when the option is enabled.
 * Added mcid_send option to allow sending a MCID request on a span.

Calendaring
--------------------------
 * Added setvar option to calendar.conf to allow setting channel variables on
   notification channels.
 * Added "calendar show types" CLI command to list registered calendar
   connectors.

MixMonitor
--------------------------
 * Added two new options, r and t with file name arguments to record 
   single direction (unmixed) audio recording separate from the bidirectional
   (mixed) recording.  The mixed file name argument is optional now as long
   as at least one recording option is used.

FollowMe
--------------------------
 * Added a new option, l, which will disable local call optimization for
   channels involved with the FollowMe thread.  Use this option to improve
   compatability for a FollowMe call with certain dialplan apps, options, and
   functions.

Meetme
--------------------------
 * Added option "k" that will automatically close the conference when there's
   only one person left when a user exits the conference.

CEL
--------------------------
 * cel_pgsql now supports the 'extra' column for data added using the
   CELGenUserEvent() application.

pbx_lua
--------------------------
 * Support for defining hints has been added to pbx_lua.  See the 'hints' table
   in the sample extensions.lua file for syntax details.
 * Applications that perform jumps in the dialplan such as Goto will now
   execute properly.  When pbx_lua detects that the context, extension, or
   priority we are executing on has changed it will immediately return control
   to the asterisk PBX engine.  Currently the engine cannot detect a Goto to
   the priority after the currently executing priority.
 * An autoservice is now started by default for pbx_lua channels.  It can be
   stopped and restarted using the autoservice_stop() and autoservice_start()
   functions.

res_fax
--------------------------
 * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
   into a FAXStatus event with an 'Operation' header that will be either
   'send', 'receive', and 'gateway'.
 * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
   Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
   feature will handle converting a fax call between an audio T.30 fax terminal
   and an IFP T.38 fax terminal.

SIP Changes
-----------
 * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
 * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
 * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.

Queue changes
-------------
 * Added general option negative_penalty_invalid default off. when set
   members are seen as invalid/logged out when there penalty is negative.
   for realtime members when set remove from queue will set penalty to -1.
 * Added queue option autopausedelay when autopause is enabled it will be
   delayed for this number of seconds since last successful call if there
   was no prior call the agent will be autopaused immediately.
 * Added member option ignorebusy this when set and ringinuse is not
   will allow per member control of multiple calls as ringinuse does for
   the Queue.
 * Added global option check_state_unknown to enforce checking of device state
   when the device state is unknown app_queue will see unknown as available.

Applications
------------
 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
   a MeetMe conference
 * Added 'k' option to MeetMe to automatically kill the conference when there's only
   one participant left (much like a normal call bridge)
 * Added extra argument to Originate to set timeout.

Asterisk Database
-----------------
 * The internal Asterisk database has been switched from Berkeley DB 1.86 to
   SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
   utility in the UTILS section of menuselect. If an existing astdb is found and no
   astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
   convert an existing astdb to the SQLite3 version automatically at runtime.

Asterisk Modules
----------------
 * Modules marked as deprecated are no longer marked as building by default. Enabling
   these modules is still available via menuselect.

IAX2 Changes
------------
 * authdebug is now disabled by default. To enable this functionaility again
   set authdebug = yes in iax.conf.

RTP Changes
-----------
 * The rtp.conf setting "strictrtp" is now enabled by default. In previous
   releases it was disabled.

PBX Core
--------
 * The PBX core previously made a call with a non-existing extension test for
   extension s@default and jump there if the extension existed.
   This was a bad default behaviour and violated the principle of least surprise.
   It has therefore been changed in this release. It may affect some
   applications and configurations that rely on this behaviour. Most channel
   drivers have avoided this for many releases by testing whether the extension
   called exists before starting the PBX and generating a local error.
   This behaviour still exists and works as before.

   Extension "s" is used when no extension is given in a channel driver,
   like immediate answer in DAHDI or calling to a domain with no user part
   in a SIP uri.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
------------------------------------------------------------------------------

SIP Changes
-----------
 * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
   now defaults to force_rport. It is very important that phones requiring nat=no be
   specifically set as such instead of relying on the default setting. If at all
   possible, all devices should have nat settings configured in the general section as
   opposed to configuring nat per-device.
 * Added preferred_codec_only option in sip.conf. This feature limits the joint
   codecs sent in response to an INVITE to the single most preferred codec.
 * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
   to be used for the outgoing call. It must be one of the codecs configured
   for the device.
 * Added tlsprivatekey option to sip.conf.  This allows a separate .pem file
   to be used for holding a private key.  If tlsprivatekey is not specified,
   tlscertfile is searched for both public and private key.
 * Added tlsclientmethod option to sip.conf.  This allows the protocol for
   outbound client connections to be specified.
 * The sendrpid parameter has been expanded to include the options
   'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
   header to be sent (equivalent to setting sendrpid=yes) and setting
   sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
 * The 'ignoresdpversion' behavior has been made automatic when the SDP received
   is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
   since the call will fail if Asterisk does not process the incoming SDP, Asterisk
   will accept the SDP even if the SDP version number is not properly incremented,
   but will generate a warning in the log indicating that the SIP peer that sent
   the SDP should have the 'ignoresdpversion' option set.
 * The 'nat' option has now been been changed to have yes, no, force_rport, and
   comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
   symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
   remote side requests it and disables symmetric RTP support. Setting it to
   force_rport forces RFC 3581 behavior and disables symmetric RTP support.
   Setting it to comedia enables RFC 3581 behavior if the remote side requests it
   and enables symmetric RTP support.
 * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
   response.  This permits the master channel to know how each channel dialled
   in a multi-channel setup resolved in an individual way. This carries a
   performance penalty and can be disabled in sip.conf using the
   'storesipcause' option.
 * Added 'externtcpport' and 'externtlsport' options to allow custom port
   configuration for the externip and externhost options when tcp or tls is used.
 * Added support for message body (stored in content variable) to SIP NOTIFY message
   accessible via AMI and CLI.
 * Added 'media_address' configuration option which can be used to explicitly specify
   the IP address to use in the SDP for media (audio, video, and text) streams.
 * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
   that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
   received.
 * Added 'use_q850_reason' configuration option for generating and parsing
   if available  Reason: Q.850;cause=<cause code> header. It is implemented
   in some gateways for better passing PRI/SS7 cause codes via SIP.
 * When dialing SIP peers, a new component may be added to the end of the dialstring
   to indicate that a specific remote IP address or host should be used when dialing
   the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
 * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
   ability to selectively force bridged channels to also be encrypted is also
   implemented. Branching in the dialplan can be done based on whether or not
   a channel has secure media and/or signaling.
 * Added directmediapermit/directmediadeny to limit which peers can send direct media
   to each other
 * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
   Charge messages to snom phones.
 * Added support for G.719 media streams.
 * Added support for 16khz signed linear media streams.
 * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
   RTP has been outfitted with the same abilities.
 * Added support for setting the Max-Forwards: header in SIP requests. Setting is
   available in device configurations as well as in the dial plan.
 * Addition of the 'subscribe_network_change' option for turning on and off
   res_stun_monitor module support in chan_sip.
 * Addition of the 'auth_options_requests' option for turning on and off
   authentication for OPTIONS requests in chan_sip.

Configuration files
-------------------
 * Add #tryinclude statement for config files.  This provides the same
   functionality as the #include statement however an asterisk module will
   still load if the filename does not exist.  Using the #include statement
   Asterisk will not allow the module to load.

IAX2 Changes
-----------
 * Added rtsavesysname option into iax.conf to allow the systname to be saved
   on realtime updates.
 * Added the ability for chan_iax2 to inform the dialplan whether or not
   encryption is being used. This interoperates with the SIP SRTP implementation
   so that a secure SIP call can be bridged to a secure IAX call when the
   dialplan requires bridged channels to be "secure".
 * Addition of the 'subscribe_network_change' option for turning on and off
   res_stun_monitor module support in chan_iax.


MGCP Changes
------------
 * Added ability to preset channel variables on indicated lines with the setvar
   configuration option.  Also, clearvars=all resets the list of variables back
   to none.
 * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
   See configs/res_pktccops.conf for more information.

XMPP Google Talk/Jingle changes
-------------------------------
  * Added the externip option to gtalk.conf.
  * Added the stunaddr option to gtalk.conf which allows for the automatic
    retrieval of the external ip from a stun server.

Applications
------------
 * Added 'p' option to PickupChan() to allow for picking up channel by the first
   match to a partial channel name.
 * Added .m3u support for Mp3Player application.
 * Added progress option to the app_dial D() option.  When progress DTMF is
   present, those values are sent immediately upon receiving a PROGRESS message
   regardless if the call has been answered or not.
 * Added functionality to the app_dial F() option to continue with execution
   at the current location when no parameters are provided.
 * Added the 'a' option to app_dial to answer the calling channel before any
   announcements or macros are executed.
 * Modified app_dial to set answertime when the called channel answers even if
   the called channel hangs up during playback of an announcement.
 * Modified app_dial 'r' option to support an additional parameter to play an
   indication tone from indications.conf
 * Added c() option to app_chanspy. This option allows custom DTMF to be set
   to cycle through the next available channel.  By default this is still '*'.
 * Added x() option to app_chanspy.  This option allows DTMF to be set to
   exit the application.
 * The Voicemail application has been improved to automatically ignore messages
   that only contain silence.
 * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
   associated mailbox(es) to be greetings-only.
 * The ChanSpy application now has the 'S' option, which makes the application
   automatically exit once it hits a point where no more channels are available