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 1.6.2 to Asterisk 1.6.3  -------------
------------------------------------------------------------------------------

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.

IAX2 Changes
-----------
 * Added rtsavesysname option into iax.conf to allow the systname to be saved
   on realtime updates.

Applications
------------
 * 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 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.
 * 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.

Dialplan Functions
------------------
 * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
   setting various connected line and redirecting party information.
 * The CHANNEL() function now supports the "name" option.
 * 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.
 * 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)

Queue changes
-------------
  * 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.

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 Reverse Charging Indication receipt & transmission (requires latest
   LibPRI).

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.

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 officianados.
   linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
   etc are performed. Thus the peices 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.

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 are supported (Exchange support
   only tested on Exchange Server 2003 with no support for forms-based authentication).

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.

------------------------------------------------------------------------------
--- 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 subscribing to a voice mailbox on a remote server and
   making the new/old message count available to local devices.
 * 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.
 * Added a new 'faxdetect=yes|no' configuration option to sip.conf.  When this
   option is enabled, a SIP channel will go to the fax extension (if it exists)
   after T38 is negotiated.  This option is disabled by default.
 * 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
------------------
 * 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 I have no idea how to use, but Leif Madsen wanted it.
   Russell says it's, like, a scope resolution function for LOCAL variables.
   Totally.  Hopefully, that means more to you than it does to me.
 * 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
------------
 * DAHDISendCallreroutingFacility parameters are now comma-separated,
   instead of the old pipe.
 * 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.
 * res_jabber: autoprune has been disabled by default, to avoid misconfiguration 
   that would end up being interpreted as a bug once Asterisk started removing 
   the contacts from a user list.
 * 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.

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

------------------------------------------------------------------------------
--- 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
    alphabetical order, as opposed to the order they were in, in the config 
    file or database. 
 * Videosupport now supports an additional option, "always", which always sets
    up video RTP ports, even on clients that don't support it.  This helps with
    callfiles and certain transfers to ensure that if two video phones are
    connected, they will always share video feeds.

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

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

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

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

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

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

Realtime changes
----------------
  * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
    adaptive capabilities.  What this means in practical terms is that if your
    realtime table lacks critical fields, Asterisk will now emit warnings to
    that effect.  Also, some of the realtime drivers have the ability (if
    configured) to automatically add those columns to the table with the
    correct type and length.

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

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

AMI - The manager (TCP/TLS/HTTP)
--------------------------------
  * Manager has undergone a lot of changes, all of them documented
    in doc/manager_1_1.txt
  * Manager version has changed to 1.1
  * Added a new action 'CoreShowChannels' to list currently defined channels
     and some information about them. 
  * Added a new action 'SIPshowregistry' to list SIP registrations.
  * Added TLS support for the manager interface and HTTP server
  * Added the URI redirect option for the built-in HTTP server
  * The output of CallerID in Manager events is now more consistent.
     CallerIDNum is used for number and CallerIDName for name.
  * Enable https support for builtin web server.
     See configs/http.conf.sample for details.
  * Added a new action, GetConfigJSON, which can return the contents of an
     Asterisk configuration file in JSON format.  This is intended to help
     improve the performance of AJAX applications using the manager interface
     over HTTP.
  * SIP and IAX manager events now use "ChannelType" in all cases where we 
     indicate channel driver. Previously, we used a mixture of "Channel"
     and "ChannelDriver" headers.
  * Added a "Bridge" action which allows you to bridge any two channels that
     are currently active on the system.
  * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
     the voicemail users setup.
  * Added 'DBDel' and 'DBDelTree' manager commands.
  * cdr_manager now reports events via the "cdr" level, separating it from
     the very verbose "call" level.
  * Manager users are now stored in memory. If you change the manager account
    list (delete or add accounts) you need to reload manager.
  * Added Masquerade manager event for when a masquerade happens between
     two channels.
  * Added "manager reload" command for the CLI
  * Lots of commands that only provided information are now allowed under the
     Reporting privilege, instead of only under Call or System.
  * The IAX* commands now require either System or Reporting privilege, to
     mirror the privileges of the SIP* commands.
  * Added ability to retrieve list of categories in a config file.
  * Added ability to retrieve the content of a particular category.
  * Added ability to empty a context.
  * Created new action to create a new file.
  * Updated delete action to allow deletion by line number with respect to category.
  * Added new action insert to add new variable to category at specified line.
  * Updated action newcat to allow new category to be inserted in file above another
    existing category.
  * Added new event "JitterBufStats" in the IAX2 channel
  * Originate now requires the Originate privilege and, if you want to call out
    to a subshell, it requires the System privilege, as well.  This was done to
    enhance manager security.
  * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264" 
  * New command: Atxfer. See doc/manager_1_1.txt for more details or 
    manager show command Atxfer from the CLI
  * New command: IAXregistry. See doc/manager_1_1.txt for more details or
    manager show command IAXregistry from the CLI

Dialplan functions
------------------
  * Added the DEVICE_STATE() dialplan function which allows retrieving any device
     state in the dialplan, as well as creating custom device states that are
     controllable from the dialplan.
  * Extend CALLERID() function with "pres" and "ton" parameters to
     fetch string representation of calling number presentation indicator
     and numeric representation of type of calling number value.
  * MailboxExists converted to dialplan function
  * A new option to Dial() for telling IP phones not to count the call
     as "missed" when dial times out and cancels.
  * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
     mutex.  No deadlocks are possible, as LOCK() only allows a single lock to be
     held for any given channel.  Also, locks are automatically freed when a
     channel is hung up.
  * Added HINT() dialplan function that allows retrieving hint information.
     Hints are mappings between extensions and devices for the sake of 
     determining the state of an extension.  This function can retrieve the list
     of devices or the name associated with a hint.
  * Added EXTENSION_STATE() dialplan function which allows retrieving the state
    of any extension.
  * Added SYSINFO() dialplan function which allows retrieval of system information
  * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
     the existence of a dialplan target.
  * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
     upper and lower case, respectively.
  * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
     ID for the call (not the Asterisk call ID or unique ID), provided that the
     channel driver supports this. For SIP, you get the SIP call-ID for the
     bridged channel which you can store in the CDR with a custom field.

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

SIP changes
-----------
  * Improved NAT and STUN support.
     chan_sip now can use port numbers in bindaddr, externip and externhost
     options, as well as contact a STUN server to detect its external address
     for the SIP socket. See sip.conf.sample, 'NAT' section.
  * The default SIP useragent= identifier now includes the Asterisk version
  * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
     If set, and the incoming request carries authentication info,
     the username to match in the users list is taken from the Digest header
     rather than from the From: field. This feature is considered experimental.
  * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
     since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
  * The "localmask" setting was removed in version 1.2 and the reminder about it
     being removed is now also removed.
  * A new option "busylevel" for setting a level of calls where asterisk reports
     a device as busy, to separate it from call-limit. This value is also added
     to the SIP_PEER dialplan function.
  * A new realtime family called "sipregs" is now supported to store SIP registration
     data. If this family is defined, "sippeers" will be used for configuration and
     "sipregs" for registrations. If it's not defined, "sippeers" will be used for
     registration data, as before.
  * The SIPPEER function have new options for port address, call and pickup groups
  * Added support for T.140 realtime text in SIP/RTP
  * The "checkmwi" option has been removed from sip.conf, as it is no longer
     required due to the restructuring of how MWI is handled.  See the descriptions 
     in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf 
     for more information.
  * Added rtpdest option to CHANNEL() dialplan function.
  * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
  * SIP now adds a header to the CANCEL if the call was answered by another phone
     in the same dial command, or if the new c option in dial() is used.
  * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
     states it is not needed. For phones, however, that do require it the "registertrying" option
     has been added so it can be enabled. 
  * A new option called "callcounter" (global/peer/user level) enables call counters needed
     for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
     used to enable this functionality).
  * New settings for timer T1 and timer B on a global level or per device. This makes it 
     possible to force timeout faster on non-responsive SIP servers. These settings are
     considered advanced, so don't use them unless you have a problem.
  * Added a dial string option to be able to set the To: header in an INVITE to any
     SIP uri.
  * Added a new global and per-peer option, qualifyfreq, which allows you to configure
     the qualify frequency.
  * Added SIP Session Timers support (RFC 4028).  This prevents stuck SIP sessions that
     were not properly torn down due to network or endpoint failures during an established
     SIP session.
  * Added experimental TCP and TLS support for SIP.  See doc/siptls.txt and 
     configs/sip.conf.sample for more information on how it is used.
  * Added a new configuration option "authfailureevents" that enables manager events when
    a peer can't authenticate properly. 
  * Added DNS manager support to registrations for peers not referencing a peer entry.

IAX2 changes
------------
  * Added the trunkmaxsize configuration option to chan_iax2.
  * Added the srvlookup option to iax.conf
  * Added support for OSP.  The token is set and retrieved through the CHANNEL()
     dialplan function.

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

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

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

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

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

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

Local channel changes
---------------------
  * The device state functionality in the Local channel driver has been updated
     to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
     to just UNKNOWN if the extension exists.
  * Added jitterbuffer support for chan_local.  This allows you to use the
     generic jitterbuffer on incoming calls going to Asterisk applications.
     For example, this would allow you to use a jitterbuffer for an incoming
     SIP call to Voicemail by putting a Local channel in the middle.  This
     feature is enabled by using the 'j' option in the Dial string to the Local
     channel in conjunction with the existing 'n' option for local channels.
  * A 'b' option has been added which causes chan_local to return the actual channel
     that is behind it when queried. This is useful for transfer scenarios as the
     actual channel will be transferred, not the Local channel.

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

DAHDI channel driver (chan_dahdi) Changes
----------------------------------------
  * SS7 support (via libss7 library)
  * In India, some carriers transmit CID via dtmf. Some code has been added
     that will handle some situations. The cidstart=polarity_IN choice has been added for
     those carriers that transmit CID via dtmf after a polarity change.
  * CID matching information is now shown when doing 'dialplan show'.
  * Added dahdi show version CLI command.
  * Added setvar support to chan_dahdi.conf channel entries.
  * Added two new options: mwimonitor and mwimonitornotify.  These options allow
     you to enable MWI monitoring on FXO lines.  When the MWI state changes,
     the script specified in the mwimonitornotify option is executed.  An internal
     event indicating the new state of the mailbox is also generated, so that
     the normal MWI facilities in Asterisk work as usual.
  * Added signalling type 'auto', which attempts to use the same signalling type
     for a channel as configured in DAHDI. This is primarily designed for analog
     ports, but will also work for digital ports that are configured for FXS or FXO
     signalling types. This mode is also the default now, so if your chan_dahdi.conf
     does not specify signalling for a channel (which is unlikely as the sample
     configuration file has always recommended specifying it for every channel) then
     the 'auto' mode will be used for that channel if possible.
  * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
     state for a channel; also ensured that the DNDState Manager event is
     emitted no matter how the DND state is set or cleared.

New Channel Drivers
-------------------
  * Added a new channel driver, chan_unistim.  See doc/unistim.txt and
     configs/unistim.conf.sample for details.  This new channel driver allows
     you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
  * Added a new channel driver, chan_console, which uses portaudio as a cross
     platform audio interface.  It was written as a channel driver that would
     work with Mac CoreAudio, but portaudio supports a number of other audio
     interfaces, as well. Note that this channel driver requires v19 or higher
     of portaudio; older versions have a different API.
 
DUNDi changes
-------------
  * Added the ability to specify arguments to the Dial application when using
     the DUNDi switch in the dialplan.
  * Added the ability to set weights for responses dynamically.  This can be
     done using a global variable or a dialplan function.  Using the SHELL()
     function would allow you to have an external script set the weight for
     each response.
  * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT.  These
     functions will allow you to initiate a DUNDi query from the dialplan,
     find out how many results there are, and access each one.

ENUM changes
------------
  * Added two new dialplan functions, ENUMQUERY and ENUMRESULT.  These
     functions will allow you to initiate an ENUM lookup from the dialplan,
     and Asterisk will cache the results.  ENUMRESULT can be used to access
     the results without doing multiple DNS queries.

Voicemail Changes
-----------------
  * Added the ability to customize which sound files are used for some of the
     prompts within the Voicemail application by changing them in voicemail.conf
  * Added the ability for the "voicemail show users" CLI command to show users
     configured by the dynamic realtime configuration method.
  * MWI (Message Waiting Indication) handling has been significantly
     restructured internally to Asterisk.  It is now totally event based
     instead of polling based.  The voicemail application will notify other
     modules that have subscribed to MWI events when something in the mailbox
     changes.
    This also means that if any other entity outside of Asterisk is changing
     the contents of mailboxes, then the voicemail application still needs to
     poll for changes.  Examples of situations that would require this option
     are web interfaces to voicemail or an email client in the case of using
     IMAP storage.  So, two new options have been added to voicemail.conf
     to account for this: "pollmailboxes" and "pollfreq".  See the sample
     configuration file for details.
  * Added "tw" language support
  * Added support for storage of greetings using an IMAP server
  * Added ability to customize forward, reverse, stop, and pause keys for message playback
  * SMDI is now enabled in voicemail using the smdienable option.
  * A "lockmode" option has been added to asterisk.conf to configure the file
     locking method used for voicemail, and potentially other things in the
     future.  The default is the old behavior, lockfile.  However, there is a
     new method, "flock", that uses a different method for situations where the
     lockfile will not work, such as on SMB/CIFS mounts.
  * Added the ability to backup deleted messages, to ease recovery in the case
     that a user accidentally deletes a message, and discovers that they need it.
  * Reworked the SMDI interface in Asterisk.  The new way to access SMDI information
     is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG().  The file
     smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
     voicemail boxes.  The SMDI interface can also poll for MWI changes when some
     outside entity is modifying the state of the mailbox (such as IMAP storage or
     a web interface of some kind).
  * Added the support for marking messages as "urgent." There are two methods to accomplish
     this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
     is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
     the message as urgent after he has recorded a voicemail by following the voice instructions.
    When listening to voicemails using VoiceMailMain urgent messages will be presented before other
     messages

Queue changes
-------------
  * Added the general option 'shared_lastcall' so that member's wrapuptime may be
     used across multiple queues.
  * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and 
     setqueueentryvar options for each queue, see queues.conf.sample for details.
  * Added keepstats option to queues.conf which will keep queue
     statistics during a reload.
  * setinterfacevar option in queues.conf also now sets a variable
     called MEMBERNAME which contains the member's name.
  * Added 'Strategy' field to manager event QueueParams which represents
     the queue strategy in use. 
  * Added option to run macro when a queue member is connected to a caller, 
     see queues.conf.sample for details.
  * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
     does not count paused queue members as unavailable.
  * Added min-announce-frequency option to queues.conf which allows you to control the
     minimum amount of time between queue announcements for use when the caller's queue
     position changes frequently.
  * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
     queue log.
  * Added ability for non-realtime queues to have realtime members
  * Added the "linear" strategy to queues.
  * Added the "wrandom" strategy to queues.
  * Added new channel variable QUEUE_MIN_PENALTY
  * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
     rules in queuerules.conf. See configs/queuerules.conf.sample for details
  * Added a new parameter for member definition, called state_interface. This may be
    used so that a member may be called via one interface but have a different interface's
    device state reported.
  * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
    "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
    "manager show command QueueReset."
  * New configuration option: randomperiodicannounce. If a list of periodic announcements is
    specified by the periodic-announce option, then one will be chosen randomly when it is time
    to play a periodic announcment
  * New configuration options: announce-position now takes two more values in addition to "yes" and
    "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
    announce-position-limit. By setting announce-position to "limit" callers will only have their
    position announced if their position is less than what is specified by announce-position-limit.
    If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
    will be told that their are more than announce-position-limit callers waiting.
  * Two new queue log events have been added. An ADDMEMBER event will be logged
    when a realtime queue member is added and a REMOVEMEMBER event will be logged
    when a realtime queue member is removed. Since there is no calling channel associated
    with these events, the string "REALTIME" is placed where the channel's unique id