CHANGES

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

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

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

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

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.

Voicemail
------------------
 * Addition of the VM_INFO function - see Dialplan function changes
 * The imapserver, imapport, and imapflags configuration options can now be
   overriden on a user by user basis.

SIP Changes
-----------
 * Asterisk will no longer substitute CID number for CID name into 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.

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

Codec changes
-------------
 * 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.

Music On Hold Changes
---------------------
 * 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).

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

Voicemail changes
-----------------
 * 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/

Applications
------------
 * 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 beneftis eg. format can
   now be used without specifying time zone as well.

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.

 * 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.

CDR postgresql driver changes
-----------------------------
 * Added command "cdr show pgsql status" to check connection status

AMI (Asterisk Manager Interface) changes
----------------------------------------
 * 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
   on option to close specific MixMonitors.

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

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

Dialplan functions
------------------
 * 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.

Followme changes
-------------
 * 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.

RTP changes
-------------
 * 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.

Text Messaging
--------------
 * 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.

------------------------------------------------------------------------------
--- 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.
 * Added 'D' command to ExternalIVR full details in doc/externalivr.txt
 * 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 doc/CCSS_architecture.pdf and doc/tex/ccss.tex(asterisk.pdf) 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
   "Security Events" chapter of the included documentation - doc/tex/asterisk.pdf.

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 doc/distributed_devstate-XMPP.txt
 * 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.

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.