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 1.8 to Asterisk 1.10 -----------------
------------------------------------------------------------------------------

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.

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.

CODECS
--------------------------
 * Ability to define custom SILK formats in codecs.conf.
 * Addition of speex32 audio format with translation.

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.

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.

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.

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.

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

SIP Changes
-----------
 * 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.
 * 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.


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
 * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
   a MeetMe conference

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 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-peroidic-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.

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.
 * 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, see doc/distributed_devstate.txt.

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 doc/externalivr.txt for the
   complete documentation.
 * 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