chan_sip: Add SubscribeContext field to SIPshowpeer AMI response

[asterisk/asterisk.git] / 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 11 to Asterisk 12 --------------------
------------------------------------------------------------------------------

AMI (Asterisk Manager Interface)
------------------
 * The SIPshowpeer action will now include a 'SubscribeContext' field for a peer
   in its response if the peer has a subscribe context set.

 * The SIPqualifypeer action now acknowledges the request once it has established
   that the request is against a known peer. It also issues a new event,
   'SIPqualifypeerdone', once the qualify action has been completed.

 * The PlayDTMF action now supports an optional 'Duration' parameter.  This
   specifies the duration of the digit to be played, in milliseconds.

 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
   updates when changes occur instead of requiring the use of pollmailboxes.

 * CLI Command 'Manager Show Commands' no longer truncates command names longer
   than 15 characters and no longer shows authorization requirement for commands.
   'Manager Show Command' now displays the privileges needed for using a given
   manager command instead.

Logging
-------------------
 * When performing queue pause/unpause on an interface without specifying an
   individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
   least one member of any queue exists for that interface.

Queue
-------------------
 * Add queue available hint.  exten => 8501,hint,Queue:markq_avail
   Note: the suffix '_avail' after the queuename.
   Reports 'InUse' for no logged in agents or no free agents.
   Reports 'Idle' when an agent is free.

Core
------------------
 * Redirecting reasons can now be set to arbitrary strings. This means
   that the REDIRECTING dialplan function can be used to set the redirecting
   reason to any string. It also allows for custom strings to be read as the
   redirecting reason from SIP Diversion headers.

------------------------------------------------------------------------------
--- 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.


MySQL
-------------------
 * The connect action in app_mysql now allows you to specify a port number to
   connect to.  This is useful if you run a MySQL server on a non-standard
   port number.


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.

 * Add queue monitoring hints.  exten => 8501,hint,Queue:markq.

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.

 * Added the AMI_CLIENT function to make manager account attributes available
   to the dialplan. It currently supports returning the current number of
   active sessions for a given account.

 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
   and the REDIRECTING functions.


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.

 * Add options namedcallgroup and namedpickupgroup to support installations
   where a higher number of groups (>64) is required.

 * Added support to use private party ID information with PRI calls.


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.

 * Add options subminexpiry and submaxexpiry to set limits of subscription
   timer independently from registration timer settings. The setting of the
   registration timer limits still is done by options minexpiry, maxexpiry
   and defaultexpiry. For backwards compatibility the setting of minexpiry
   and maxexpiry also is used to configure the subscription timer limits if
   subminexpiry and submaxexpiry are not set in sip.conf.

 * Set registration timer limits to default values when reloading sip
   configuration and values are not set by configuration.

 * Add options namedcallgroup and namedpickupgroup to support installations
   where a higher number of groups (>64) is required.

 * 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.

 * Extra headers specified with SIPAddHeader are sent with the REFER message
   when using Transfer application. See refer_addheaders in sip.conf.sample.

 * Added support to use private party ID information with calls.


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


chan_mISDN:
------------------
 * Add options namedcallgroup and namedpickupgroup to support installations
   where a higher number of groups (>64) is required.

 * Added support to use private party ID information with calls.


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.

 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
   items (separated by commas), and items in the rule can be negated by prefixing
   them with '!'. This simplifies Asterisk Realtime configurations, since it is no
   longer necessray to control the order that the 'permit' and 'deny' columns are
   returned from queries.

 * 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.

 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
   is available.

 * Call files now support the "early_media" option to connect with an outgoing
   extension when early media is received.

 * Added support to use private party ID information with calls.


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)
------------------
 * The originate action now has an option "EarlyMedia" that enables the
   call to bridge when we get early media in the call. Previously,
   early media was disregarded always when originating calls using AMI.

 * Added setvar= option to manager accounts (much like sip.conf)

 * 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.

 * Added SIPpeerstatus manager command which will generate PeerStatus events
   similar to the existing PeerStatus events found in chan_sip on demand.

 * 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 "dialplan add include" command has been modified to create context a context
   if one does not already exist. For instance, "dialplan add include foo into bar"
   will create context "bar" if it does not already exist.

 * A  "dialplan remove context" command has been added to remove a context from
   the dialplan

 * 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.

 * The iLBC codec can now use a system-provided iLBC library if one is installed,
   just like the GSM codec.

DUNDi changes
-------------
 * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
   'DONTASK' hints in the cache and list all DUNDi cache entires respectively.

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
   to spy on.
 * The ChanSpy application also now has the 'E' option, which spies on a single
   channel and exits when that channel hangs up.
 * The MeetMe application now turns on the DENOISE() function by default, for
   each participant.  In our tests, this has significantly decreased background
   noise (especially noisy data centers).
 * Voicemail now permits storage of secrets in a separate file, located in the
   spool directory of each individual user.  The control for this is located in
   the "passwordlocation" option in voicemail.conf.  Please see the sample
   configuration for more information.
 * The ChanIsAvail application now exposes the returned cause code using a separate
   variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
 * Added 'd' option to app_followme.  This option disables the "Please hold"
   announcement.
 * Added 'y' option to app_record. This option enables a mode where any DTMF digit
   received will terminate recording.
 * Voicemail now supports per mailbox settings for folders when using IMAP storage.
   Previously the folder could only be set per context, but has now been extended
   using the imapfolder option.
 * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
 * Voicemail now allows the pager date format to be specified separately from the
   email date format.
 * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
   to allow joining, leaving, and sending text to group chats.
 * MeetMe has a new option 'G' to play an announcement before joining a conference.
 * Page has a new option 'A(x)' which will playback an announcement simultaneously
   to all paged phones (and optionally excluding the caller's one using the new
   option 'n') before the call is bridged.
 * The 'f' option to Dial has been augmented to take an optional argument. If no
   argument is provided, the 'f' option works as it always has. If an argument is
   provided, then the connected party information of all outgoing channels created
   during the Dial will be set to the argument passed to the 'f' option.
 * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
   Gosub on the peer.
 * The OSP lookup application adds in/outbound network ID, optional security,
   number portability, QoS reporting, destination IP port, custom info and service
   type features.
 * Added new application VMSayName that will play the recorded name of the voicemail
   user if it exists, otherwise will play the mailbox number.
 * Added custom device states to ConfBridge bridges.  Use 'confbridge:<name>' to
   retrieve state for a particular bridge, where <name> is the conference name
 * app_directory now allows exiting at any time using the operator or pound key.
 * Voicemail now supports setting a locale per-mailbox.
 * Two new applications are provided for declining counting phrases in multiple
   languages.  See the application notes for SayCountedNoun and SayCountedAdj for
   more information.
 * Voicemail now runs the externnotify script when pollmailboxes is activated and
   notices a change.
 * Voicemail now includes rdnis within msgXXXX.txt file.
 * ExternalIVR now supports IPv6 addresses.
 * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
   at https://wiki.asterisk.org/wiki/x/oQBB
 * ParkedCall and Park can now specify the parking lot to use.

Dialplan Functions
------------------
 * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
   over SRV records associated with a specific service. From the CLI, type
   'core show function SRVQUERY' and 'core show function SRVRESULT' for more
   details on how these may be used.
 * PITCH_SHIFT dialplan function added. This function can be used to modify the
   pitch of a channel's tx and rx audio streams.
 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
   setting various connected line and redirecting party information.
 * CALLERID and CONNECTEDLINE dialplan functions have been extended to
   support ISDN subaddressing.
 * The CHANNEL() function now supports the "name" and "checkhangup" options.
 * For DAHDI channels, the CHANNEL() dialplan function now allows
   the dialplan to request changes in the configuration of the active
   echo canceller on the channel (if any), for the current call only.
   The syntax is:

   exten => s,n,Set(CHANNEL(echocan_mode)=off)

   The possible values are:

     on - normal mode (the echo canceller is actually reinitialized)
     off - disabled
     fax - FAX/data mode (NLP disabled if possible, otherwise completely
           disabled)
     voice - voice mode (returns from FAX mode, reverting the changes that
             were made when FAX mode was requested)
 * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
   and setting variables on the channel which created the current channel.
   Administrators should take care to avoid naming conflicts, when multiple
   channels are dialled at once, especially when used with the Local channel
   construct (which all could set variables on the master channel).  Usage
   of the HASH() dialplan function, with the key set to the name of the slave
   channel, is one approach that will avoid conflicts.
 * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
   audio in a channel.
 * func_odbc now allows multiple row results to be retrieved without using
   mode=multirow.  If rowlimit is set, then additional rows may be retrieved
   from the same query by using the name of the function which retrieved the
   first row as an argument to ODBC_FETCH().
 * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
   dialplan. This function returns the content of the received message.
 * Added REPLACE, which searches a given variable name for a set of characters,
   then either replaces them with a single character or deletes them.
 * Added PASSTHRU, which literally passes the same argument back as its return
   value.  The intent is to be able to use a literal string argument to
   functions that currently require a variable name as an argument.
 * HASH-associated variables now can be inherited across channel creation, by
   prefixing the name of the hash at assignment with the appropriate number of
   underscores, just like variables.
 * GROUP_MATCH_COUNT has been improved to allow regex matching on category
 * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
   whether or not channels that are bridged to the current channel will be
   required to have secure signaling and/or media.
 * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
   the current channel has secure signaling and/or media.
 * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
   "no_media_path" option.
   Returns "0" if there is a B channel associated with the call.
   Returns "1" if no B channel is associated with the call.  The call is either
   on hold or is a call waiting call.
 * Added option to dialplan function CDR(), the 'f' option
   allows for high resolution times for billsec and duration fields.
 * FILE() now supports line-mode and writing.
 * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
 * FRAME_TRACE(), for tracking internal ast_frames on a channel.

Dialplan Variables
------------------
 * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
 * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
   and is set when a dynamic feature is triggered.
 * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
   to dynamically create a new parking lot matching the value this varible is
   set to.
 * Added PARKINGDYNAMIC which represents the template parkinglot defined in
   features.conf that should be the base for dynamic parkinglots.
 * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
   parkinglot should have.
 * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
   parkinglot should have.
 * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
   should have.

Queue changes
-------------
 * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
   timeout has expired.
 * Added 'R' option to app_queue.  This option stops moh and indicates ringing
   to the caller when an Agent's phone is ringing.  This can be used to indicate
   to the caller that their call is about to be picked up, which is nice when
   one has been on hold for an extened period of time.
 * A new config option, penaltymemberslimit, has been added to queues.conf.
   When set this option will disregard penalty settings when a queue has too
   few members.
 * A new option, 'I' has been added to both app_queue and app_dial.
   By setting this option, Asterisk will not update the caller with
   connected line changes or redirecting party changes when they occur.
 * A 'relative-periodic-announce' option has been added to queues.conf.  When
   enabled, this option will cause periodic announce times to be calculated
   from the end of announcements rather than from the beginning.
 * The autopause option in queues.conf can be passed a new value, "all." The
   result is that if a member becomes auto-paused, he will be paused in all
   queues for which he is a member, not just the queue that failed to reach
   the member.
 * Added dialplan function QUEUE_EXISTS to check if a queue exists
 * The queue logger now allows events to optionally propagate to a file,
   even when realtime logging is turned on.  Additionally, realtime logging
   supports sending the event arguments to 5 individual fields, although it
   will fallback to the previous data definition, if the new table layout is
   not found.

mISDN channel driver (chan_misdn) changes
----------------------------------------
 * Added display_connected parameter to misdn.conf to put a display string
   in the CONNECT message containing the connected name and/or number if
   the presentation setting permits it.
 * Added display_setup parameter to misdn.conf to put a display string
   in the SETUP message containing the caller name and/or number if the
   presentation setting permits it.
 * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
   indicate the dialplan settings are to be obtained from the asterisk
   channel.
 * Made misdn.conf parameter callerid accept the "name" <number> format
   used by the rest of the system.
 * Made use the nationalprefix and internationalprefix misdn.conf
   parameters to prefix any received number from the ISDN link if that
   number has the corresponding Type-Of-Number.  NOTE:  This includes
   comparing the incoming call's dialed number against the MSN list.
 * Added the following new parameters: unknownprefix, netspecificprefix,
   subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
   received number from the ISDN link if that number has the corresponding
   Type-Of-Number.
 * Added new dialplan application misdn_command which permits controlling
   the CCBS/CCNR functionality.
 * Added new dialplan function mISDN_CC which permits retrieval of various
   values from an active call completion record.
 * For PTP, you should manually send the COLR of the redirected-to party
   for an incomming redirected call if the incoming call could experience
   further redirects.  Just set the REDIRECTING(to-num,i) = ${EXTEN} and
   set the REDIRECTING(to-pres) to the COLR.  A call has been redirected
   if the REDIRECTING(from-num) is not empty.
 * For outgoing PTP redirected calls, you now need to use the inhibit(i)
   option on all of the REDIRECTING statements before dialing the
   redirected-to party.  You still have to set the REDIRECTING(to-xxx,i)
   and the REDIRECTING(from-xxx,i) values.  The PTP call will update the
   redirecting-to presentation (COLR) when it becomes available.
 * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
   information.

thirdparty mISDN enhancements
-----------------------------
mISDN has been modified by Digium, Inc. to greatly expand facility message
support to allow:
  * Enhanced COLP support for call diversion and transfer.
  * CCBS/CCNR support.

The latest modified mISDN v1.1.x based version is available at:
http://svn.digium.com/svn/thirdparty/mISDN/trunk
http://svn.digium.com/svn/thirdparty/mISDNuser/trunk

Tagged versions of the modified mISDN code are available under:
http://svn.digium.com/svn/thirdparty/mISDN/tags
http://svn.digium.com/svn/thirdparty/mISDNuser/tags

libpri channel driver (chan_dahdi) DAHDI changes
-------------------------------------------
 * The channel variable PRIREDIRECTREASON is now just a status variable
   and it is also deprecated.  Use the REDIRECTING(reason) dialplan function
   to read and alter the reason.
 * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
   redirected-to party for an incomming redirected call if the incoming call
   could experience further redirects.  Just set the
   REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
   to the COLR.  A call has been redirected if the REDIRECTING(count) is not
   zero.
 * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
   use the inhibit(i) option on all of the REDIRECTING statements before
   dialing the redirected-to party.  You still have to set the
   REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values.  The call
   will update the redirecting-to presentation (COLR) when it becomes available.
 * Added the ability to ignore calls that are not in a Multiple Subscriber
   Number (MSN) list for PTMP CPE interfaces.
 * Added dynamic range compression support for dahdi channels.  It is
   configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
 * Added support for ISDN calling and called subaddress with partial support
   for connected line subaddress.
 * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
 * Added handling of received HOLD/RETRIEVE messages and the optional ability
   to transfer a held call on disconnect similar to an analog phone.
 * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
   Will reroute/deflect an outgoing call when receive the message.
   Can use the DAHDISendCallreroutingFacility to send the message for the
   supported switches.
 * Added standard location to add options to chan_dahdi dialing:
   Dial(DAHDI/g1[/extension[/options]])
   Current options:
   K(<keypad_digits>)
   R Reverse charging indication
 * Added Reverse Charging Indication (Collect calls) send/receive option.
   Send reverse charging in SETUP message with the chan_dahdi R dialing option.
   Dial(DAHDI/g1/extension/R)
   Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
   (requires latest LibPRI)
 * Added ability to send/receive keypad digits in the SETUP message.
   Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
   dialing option.  Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
   Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
   (requires latest LibPRI)
 * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
   to eliminate tromboned calls.  A tromboned call goes out an interface and comes
   back into the same interface.  Tromboned calls happen because of call routing,
   call deflection, call forwarding, and call transfer.
 * Added the ability to send and receive ETSI Advice-Of-Charge messages.
 * Added the ability to support call waiting calls.  (The SETUP has no B channel
   assigned.)
 * Added Malicious Call ID (MCID) event to the AMI call event class.
 * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).

Asterisk Manager Interface
--------------------------
 * The Hangup action now accepts a Cause header which may be used to
   set the channel's hangup cause.
 * sslprivatekey option added to manager.conf and http.conf.  Adds the ability
   to specify a separate .pem file to hold a private key.  By default sslcert
   is used to hold both the public and private key.
 * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
   for options containing the 'tls' prefix.  For example, 'sslenable' is now
   'tlsenable'.  This has been done in effort to keep ssl and tls options consistent
   across all .conf files. All affected sample.conf files have been modified to
   reflect this change.  Previous options such as 'sslenable' still work,
   but options with the 'tls' prefix are preferred.
 * Added a MuteAudio AMI action for muting inbound and/or outbound audio
   in a channel. (res_mutestream.so)
 * The configuration file manager.conf now supports a channelvars option, which
   specifies a list of channel variables to include in each channel-oriented
   event.
 * The redirect command now has new parameters ExtraContext, ExtraExtension,
   and ExtraPriority to allow redirecting the second channel to a different
   location than the first.
 * Added new event "JabberStatus" in the Jabber module to monitor buddies
   status.
 * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
   in a MixMonitor recording.
 * The 'iax2 show peers' output is now similar to the expected output of
   'sip show peers'.
 * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
   aoc event class.
 * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
   AOC-E messages on a channel.
 * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
   conform more closely to similar events.
 * Added a new eventfilter option per user to allow whitelisting and blacklisting
   of events.
 * Added optional parkinglot variable for park command.
 * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
   if CallerIDNum and CallerIDName headers are also present.

Channel Event Logging
---------------------
 * A new interface, CEL, is introduced here. CEL logs single events, much like
   the AMI, but it differs from the AMI in that it logs to db backends much
   like CDR does; is based on the event subsystem introduced by Russell, and
   can share in all its benefits; allows multiple backends to operate like CDR;
   is specialized to event data that would be of concern to billing sytems,
   like CDR. Backends for logging and accounting calls have been produced,
   but a new CDR backend is still in development.

CDR
---
 * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
   linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
   etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
 * Multiple files and formats can now be specified in cdr_custom.conf.
 * cdr_syslog has been added which allows CDRs to be written directly to syslog.
   See configs/cdr_syslog.conf.sample for more information.
 * A 'sequence' field has been added to CDRs which can be combined with
   linkedid or uniqueid to uniquely identify a CDR.
 * Handling of billsec and duration field has changed. If your table definition
   specifies those fields as float,double or similar they will now be logged with
   microsecond accuracy instead of a whole integer.

Calendaring for Asterisk
------------------------
 * A new set of modules were added supporing calendar integration with Asterisk.
   Dialplan functions for reading from and writing to calendars are included,
   as well as the ability to execute dialplan logic upon calendar event notifications.
   iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
   Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
   Exchange Server 2007+ with full write and attendee support) are supported (Exchange
   2003 support does not support forms-based authentication).

Call Completion Supplementary Services for Asterisk
---------------------------------------------------
 * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
   DAHDI/ISDN supports call completion for the following switch types:
   EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
   See https://wiki.asterisk.org/wiki/x/2ABQ for details.

Multicast RTP Support
---------------------
 * A new RTP engine and channel driver have been added which supports Multicast RTP.
   The channel driver can be used with the Page application to perform multicast RTP
   paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
   Type can be either basic or linksys.
   Destination is the IP address and port for the RTP packets.
   Control address is specific to the linksys type and is used for sending the control
   packets unique to them.

Security Events Framework
-------------------------
 * Asterisk has a new C API for reporting security events.  The module res_security_log
   sends these events to the "security" logger level.  Currently, AMI is the only
   Asterisk component that reports security events.  However, SIP support will be
   coming soon.  For more information on the security events framework, see the
   "Asterisk Security Framework" section of the Asterisk wiki at
   https://wiki.asterisk.org/wiki/x/wgBQ
 * SIP support was added in Asterisk 10
 * This API now supports IPv6 addresses

Fax
---
 * A technology independent fax frontend (res_fax) has been added to Asterisk.
 * A spandsp based fax backend (res_fax_spandsp) has been added.
 * The app_fax module has been deprecated in favor of the res_fax module and
   the new res_fax_spandsp backend.
 * The SendFAX and ReceiveFAX applications now send their log messages to a
   'fax' logger level, instead of to the generic logger levels. To see these
   messages, the system's logger.conf file will need to direct the 'fax' logger
   level to one or more destinations; the logger.conf.sample file includes an
   example of how to do this. Note that if the 'fax' logger level is *not*
   directed to at least one destination, log messages generated by these
   applications will be lost, and that if the 'fax' logger level is directed to
   the console, the 'core set verbose' and 'core set debug' CLI commands will
   have no effect on whether the messages appear on the console or not.

Miscellaneous
-------------
 * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
   Now, in order to enable transmitting silence during record the transmit_silence
   option should be used.  transmit_silence_during_record remains a valid option, but
   defaults to the behavior of the transmit_silence option.
 * Addition of the Unit Test Framework API for managing registration and execution
   of unit tests with the purpose of verifying the operation of C functions.
 * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
   XMPP text messages to the remote JID.
 * Modules.conf has a new option - "require" - that marks a module as critical for
   the execution of Asterisk.
   If one of the required modules fail to load, Asterisk will exit with a return
   code set to 2.
 * An 'X' option has been added to the asterisk application which enables #exec support.
   This allows #exec to be used in asterisk.conf.
 * jabber.conf supports a new option auth_policy that toggles auto user registration.
 * A new lockconfdir option has been added to asterisk.conf to protect the
   configuration directory (/etc/asterisk by default) during reloads.
 * The parkeddynamic option has been added to features.conf to enable the creation
   of dynamic parkinglots.
 * chan_dahdi now supports reporting alarms over AMI either by channel or span via
   the reportalarms config option.
 * chan_dahdi supports dialing configuring and dialing by device file name.
   DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
   it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
 * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
   False by default. If set, chan_dahdi will ignore failed 'channel' entries.
   Handy for the above name-based syntax as it does not depend on
   initialization order.
 * The Realtime dialplan switch now caches entries for 1 second.  This provides a
   significant increase in performance (about 3X) for installations using this switchtype.
 * Distributed devicestate now supports the use of the XMPP protocol, in addition to
   AIS.  For more information, please see the Distributed Device State section of the
   Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
 * The addition of G.719 pass-through support.
 * Added support for 16khz Speex audio.  This can be enabled by using 'allow=speex16'
   during device configuration.
 * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
   have less than 3 lines on the LCD.
 * Realtime now supports database failover.  See the sample extconfig.conf for details.
 * The addition of improved translation path building for wideband codecs.  Sample
   rate changes during translation are now avoided unless absolutely necessary.
 * The addition of the res_stun_monitor module for monitoring and reacting to network
   changes while behind a NAT.
 * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
   DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
   These allow support for any Administration. Default is AT&T values.

CLI Changes
-----------
 * The 'core set debug' and 'core set verbose' commands, in previous versions, could
   optionally accept a filename, to apply the setting only to the code generated from
   that source file when Asterisk was built. However, there are some modules in Asterisk
   that are composed of multiple source files, so this did not result in the behavior
   that users expected. In this version, 'core set debug' and 'core set verbose'
   can optionally accept *module* names instead (with or without the .so extension),
   which applies the setting to the entire module specified, regardless of which source
   files it was built from.
 * New 'manager show settings' command showing the current settings loaded from
   manager.conf.
 * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
   the channel hangup request to all channels.
 * Added a "core reload" CLI command that executes a global reload of Asterisk.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2  -------------
------------------------------------------------------------------------------

SIP Changes
-----------
 * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
   Snom phones use this for call pickup of extensions that the phone is
   subscribed to.
 * Added support for setting the domain in the URI for caller of an
   outbound call by using the SIPFROMDOMAIN channel variable.
 * Added a new configuration option "remotesecret" for authentication to
   remote services. For backwards compatibility, "secret" still has the
   same function as before, but now you can configure both a remote secret and a
   local secret for mutual authentication.
 * If the channel variable  ATTENDED_TRANSFER_COMPLETE_SOUND is set,
   the sound will be played to the target of an attended transfer
 * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
   finer control over how many peers Asterisk will qualify and the gap between them
   when all peers need to be qualified at the same time.
 * Added a new 'ignoresdpversion' option to sip.conf.  When this is enabled
   (either globally or for a specific peer), chan_sip will treat any SDP data
   it receives as new data and update the media stream accordingly.  By
   default, Asterisk will only modify the media stream if the SDP session
   version received is different from the current SDP session version.  This
   option is required to interoperate with devices that have non-standard SDP
   session version implementations (observed with Microsoft OCS).  This option
   is disabled by default.
 * The parsing of register => lines in sip.conf has been modified to allow a port
   to be present in the "user" portion. Please see the sip.conf.sample file for more
   information
 * Added support for subscribing to MWI on a remote server and making the status available
   as a mailbox. Please see the sip.conf.sample file for more information.
 * Added a function to remove SIP headers added in the dialplan before the
   first INVITE is generated - SIPRemoveHeader()
 * Channel variables set with setvar= in a device configuration is now
   set both for inbound and outbound calls.
 * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.

IAX2 changes
------------
  * Added immediate option to iax.conf
  * Added forceencryption option to iax.conf
  * Added Encryption and Trunk status to manager command "iaxpeers"

Skinny Changes
--------------
 * The configuration file now holds separate sections for devices and lines.
   Please have a look at configs/skinny.conf.sample and change your skinny.conf
   accordingly.

DAHDI Changes
-------------
 * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
   support for LibOpenR2.  http://www.libopenr2.org/
 * The UK option waitfordialtone has been added for use with BT analog
   lines.
 * Added a 'faxbuffers' configuration option to chan_dahdi.conf.  This option
   is used in conjunction with the 'faxdetect' configuration option.  When
   'faxbuffers' is used and fax tones are detected, the channel will dynamically
   switch to the configured faxbuffers policy.  For example, to use 6 buffers
   and a 'full' buffer policy for a fax transmission, add:
     faxbuffers=>6,full
   The faxbuffers configuration will be in affect until the call is torn down.
 * Added service message support for 4ESS/5ESS switches.

Dialplan Functions
------------------
 * For DAHDI channels, the CHANNEL() dialplan function now
   supports changing the channel's buffer policy (for the current
   call only), using this syntax:

   exten => s,n,Set(CHANNEL(buffers)=6,full)

   This would change the channel to the 'full' buffer policy and
   6 (six) buffers. Possible options for this setting are the same
   as those in chan_dahdi.conf.
 * Added a new dialplan function, CURLOPT, which permits setting various
   options that may be useful with the CURL dialplan function, such as
   cookies, proxies, connection timeouts, passwords, etc.
 * Permit the syntax and synopsis fields of the corresponding dialplan
   functions to be individually set from func_odbc.conf.
 * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
 * func_odbc now may specify an insert query to execute, when the write query
   affects 0 rows (usually indicating that no such row exists).
 * Added a new dialplan function, LISTFILTER, which permits removing elements
   from a set list, by name.  Uses the same general syntax as the existing CUT
   and FIELDQTY dialplan functions, which also manage lists.
 * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
   obtaining realtime data from the dialplan.
 * Added LOCAL_PEEK, which allows access to variables in any stack frame within
   a subroutine when using the GoSub() and Return() applications.
 * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
   of "core show function AUDIOHOOK_INHERIT" from the CLI
 * Added AES_ENCRYPT. For information on its use, please see the output
   of "core show function AES_ENCRYPT" from the CLI
 * Added AES_DECRYPT. For information on its use, please see the output
   of "core show function AES_DECRYPT" from the CLI
 * func_odbc now supports database transactions across multiple queries.

Applications
------------
 * Scheduled meetme conferences may now have their end times extended by
   using MeetMeAdmin.
 * app_authenticate now gives the ability to select a prompt other than
   the default.
 * app_directory now pays attention to the searchcontexts setting in
   voicemail.conf and will look through all contexts, if no context is
   specified in the initial argument.
 * A new application, Originate, has been introduced, that allows asynchronous
   call origination from the dialplan.
 * Voicemail now permits setting the emailsubject and emailbody per mailbox,
   in addition to the setting in the "general" context.
 * Added ConfBridge dialplan application which does conference bridges without
   DAHDI. For information on its use, please see the output of
   "core show application ConfBridge" from the CLI.

Miscellaneous
-------------
 * The Asterisk CLI has a new command, "channel redirect", which is similar in
   operation to the AMI Redirect action.
 * extensions.conf now allows you to use keyword "same" to define an extension
   without actually specifying an extension.  It uses exactly the same pattern
   as previously used on the last "exten" line.  For example:
     exten => 123,1,NoOp(something)
     same  =>     n,SomethingElse()
 * musiconhold.conf classes of type 'files' can now use relative directory paths,
   which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
 * All deprecated CLI commands are removed from the sourcecode. They are now handled
   by the new clialiases module. See cli_aliases.conf.sample file.
 * Times within timespecs are now accurate down to the minute.  This is a change
   from historical Asterisk, which only provided timespecs rounded to the nearest
   even (read: evenly divisible by 2) minute mark.
 * The realtime switch now supports an option flag, 'p', which disables searches for
   pattern matches.
 * In addition to a time range and date range, timespecs now accept a 5th optional
   argument, timezone.  This allows you to perform time checks on alternate
   timezones, especially if those daylight savings time ranges vary from your
   machine's native timezone.  See GotoIfTime, ExecIfTime, IFTIME(), and timed
   includes.
 * The contrib/scripts/ directory now has a script called sip_nat_settings that will
   give you the correct output for an asterisk box behind nat. It will give you the
   externhost and localnet settings.
 * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
   can connect calls in passthrough mode, as well as record and play back files.
 * Successful and unsuccessful call pickup can now be alerted through sounds, by
   using pickupsound and pickupfailsound in features.conf.
 * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
   This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
   instead of the /var/run/asterisk.pid where it used to be. This will make
   installs as non-root easier to manage.

CDR
---

* The cdr.conf file must exist and be correctly programmed in order for CDR records to
  be written; they will no longer be explicitly written.

Asterisk Manager Interface
--------------------------
 * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
   a non-empty value) in your request. If you do this, any pending AMI events will
   *not* be included in the response to your request as they would normally, but
   will be left in the event queue for the next request you make to retrieve. For
   some applications, this will allow you to guarantee that you will only see
   events in responses to 'WaitEvent' actions, and can better know when to expect them.
   To know whether the Asterisk server supports this header or not, your client can
   inspect the first response back from the server to see if it includes this header:

   Pragma: SuppressEvents

   If this is included, the server supports event suppression.

 * Added 4 new Actions to list skinny device(s) and line(s)
   SKINNYdevices
   SKINNYshowdevice
   SKINNYlines
   SKINNYshowline

LDAP Schema File Additions
--------------------------
 * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox  objectClasses
   to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
 * Added new Fields:
   - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
   - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
   - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
 * Removed redundant IPaddr (there's already IPAddress)
   - Gives more configuration Flags for SIP-Users available (tested)
   - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
     without extensibleObject (which really should be the last resort); gives
     also additional possibilities for LDAP-filter

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1  -------------
------------------------------------------------------------------------------

Device State Handling
---------------------
 * The event infrastructure in Asterisk got another big update to help support
   distributed events.  It currently supports distributed device state and
   distributed Voicemail MWI (Message Waiting Indication).  A new module has
   been merged, res_ais, which facilitates communicating events between servers.
   It uses the SAForum AIS (Service Availability Forum Application Interface
   Specification) CLM (Cluster Management) and EVT (Event) services to maintain
   a cluster of Asterisk servers, and to share events between them.  For more
   information on setting this up, refer to the Distributed Device State section
   of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ

Dialplan Functions
------------------
 * Added a new dialplan function, AST_CONFIG(), which allows you to access
   variables from an Asterisk configuration file.
 * The JACK_HOOK function now has a c() option to supply a custom client name.
 * Added two new dialplan functions from libspeex for audio gain control and
   denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
   rx directions of a channel from the dialplan.
 * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
   based on other parameters.  The default is still to search based on the
   forwarding station ID.  However, there are new options that allow you to search
   based on the message desk terminal ID, or the message desk number.
 * TIMEOUT() has been modified to be accurate down to the millisecond.
 * ENUM*() functions now include the following new options:
     - 'u' returns the full URI and does not strip off the URI-scheme.
     - 's' triggers ISN specific rewriting
     - 'i' looks for branches into an Infrastructure ENUM tree
     - 'd' for a direct DNS lookup without any flipping of digits.
 * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
 * CHANNEL() now has options for the maximum, minimum, and standard or normal
   deviation of jitter, rtt, and loss for a call using chan_sip.

DAHDI channel driver (chan_dahdi) Changes
----------------------------------------
 * Channels can now be configured using named sections in chan_dahdi.conf, just
   like other channel drivers, including the use of templates.
 * The default for pridialplan has changed from 'national' to 'unknown'.

PBX Changes
-----------
 * It is now possible to specify a pattern match as a hint. Once a phone subscribes
   to something that matches the pattern a hint will be created using the contents
   and variables evaluated.
 * Dialplan matching has been extended to allow an extension to return to the
   PBX core to wait for more digits.  This is done by using the new dialplan
   application called "Incomplete".  This will permit a whole new level of
   extension control, by giving the administrator more control over early
   matches employing one of the short-circuit pattern match operators.  Note
   that custom applications can trigger this same behavior by returning the
   special value AST_PBX_INCOMPLETE.

Application Changes
-------------------
 * Directory now permits both first and last names to be matched at the same
   time.  In addition, the number of digits to enter of the name can be set in
   the arguments to Directory; previously, you could enter only 3, regardless
   of how many names are in your company.  For large companies, this should be
   quite helpful.
 * Voicemail now permits a mailbox setting to wrap around from first to last
   messages, if the "messagewrap" option is set to a true value.
 * Voicemail now permits an external script to be run, for password validation.
   The script should output "VALID" or "INVALID" on stdout, depending upon the
   wish to validate or invalidate the password given.  Arguments are:
   "mailbox" "context" "oldpass" "newpass".  See the sample voicemail.conf for
   more details
 * Dial has a new option: F(context^extension^pri), which permits a callee to
   continue in the dialplan, at the specified label, if the caller hangs up.
 * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
   technology name (e.g. SIP, IAX, etc) of the channel being spied on.
 * The Jack application now has a c() option to supply a custom client name.
 * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
   like the pre-existing whisper mode, except that the spy can also talk to the
   participant on the bridged channel as well.
 * Chanspy has a new option, 'n', which will allow for the spied-on party's name
   to be spoken instead of the channel name or number. For more information on the
   use of this option, issue the command "core show application ChanSpy" from the
   Asterisk CLI.
 * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
   spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
   words, if using the 'd' option, it is not possible to enter a number to append to
   the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
   change to whisper mode, and pressing 6 will change to barge mode.
 * ExternalIVR now takes several options that affect the way it performs, as
   well as having several new commands.  Please see the External IVR page on the Asterisk
   wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
 * Added ability to communicate over a TCP socket instead of forking a child process for the
   ExternalIVR application.
 * ChanIsAvail has a new option, 'a', which will return all available channels instead
   of just the first one if you give the function more then one channel to check.
 * PrivacyManager now takes an option where you can specify a context where the
   given number will be matched. This way you have more control over who is allowed
   and it stops the people who blindly enter 10 digits.
 * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
   answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
   from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
   original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
   the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
   obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
 * The Dial() application no longer copies the language used by the caller to the callee's
   channel. If you desire for the caller's channel's language to be used for file playback
   to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
 * SendImage() no longer hangs up the channel on error; instead, it sets the
   status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
   'UNSUPPORTED'.  This change makes SendImage() more consistent with other
   applications.
 * Park has a new option, 's', which silences the announcement of the parking space number.
 * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
   invalid input and will be assumed to mean that no timeout is desired.

SIP Changes
-----------
 * Added DNS manager support to registrations for peers referencing peer entries.
   DNS manager runs in the background which allows DNS lookups to be run asynchronously
   as well as periodically updating the IP address. These properties allow for
   better performance as well as recovery in the event of an IP change.
 * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
   load/reload of large numbers of peers/users by ~40x (for large lists of peers).
   These changes also provide performance improvements for call setup and tear down.
 * Added ability to specify registration expiry time on a per registration basis in
   the register line.
 * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
   lost packets.
 * Added t38pt_usertpsource option. See sip.conf.sample for details.
 * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
 * 'sip show peers' and 'sip show users' display their entries sorted in
    alphabetical order, as opposed to the order they were in, in the config
    file or database.
 * Videosupport now supports an additional option, "always", which always sets
    up video RTP ports, even on clients that don't support it.  This helps with
    callfiles and certain transfers to ensure that if two video phones are
    connected, they will always share video feeds.

IAX Changes
-----------
 * Existing DNS manager lookups extended to check for SRV records.
 * IAX2 encryption support has been improved to support periodic key rotation
   within a call for enhanced security.  The option "keyrotate" has been
   provided to disable this functionality to preserve backwards compatibility
   with older versions of IAX2 that do not support key rotation.

CLI Changes
-----------
  * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
     data tree based on the given <path>.
  * New CLI command "data show providers" that will display all the registered
     callbacks.
  * New CLI command, "config reload <file.conf>" which reloads any module that
     references that particular configuration file.  Also added "config list"
     which shows which configuration files are in use.
  * New CLI commands, "pri show version" and "ss7 show version" that will
     display which version of libpri and libss7 are being used, respectively.
     A new API call was added so trunk will now have to be compiled against
     a versions of libpri and libss7 that have them or it will not know that
     these libraries exist.
  * The commands "core show globals", "core set global" and "core set chanvar" has
     been deprecated in favor of the more semanticly correct "dialplan show globals",
     "dialplan set chanvar" and "dialplan set global".
  * New CLI command "dialplan show chanvar" to list all variables associated
    with a given channel.

DNS manager changes
-------------------
  * Addresses managed by DNS manager now can check to see if there is a DNS
    SRV record for a given domain and will use that hostname/port if present.

AMI - The manager (TCP/TLS/HTTP)
--------------------------------
  * The Status command now takes an optional list of variables to display
    along with channel status.
  * The QueueEntry event now also includes the channel's uniqueid

ODBC Changes
------------
  * res_odbc no longer has a limit of 1023 total possible unshared connections,
    as some people were running into this limit.  This limit has been increased
    to 4.2 billion.

Queue changes
-------------
  * The TRANSFER queue log entry now includes the the caller's original
    position in the transferred-from queue.
  * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
    "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
    as well as an explanation about timeout options in general
  * Added a new option - C - for forcing the "answered elsewhere" flag on
    cancellation of calls in to members of the queue. This is to avoid the
    call to a member of a queue having the call listed as a "missed call".

Realtime changes
----------------
  * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
    adaptive capabilities.  What this means in practical terms is that if your
    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

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.