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Creating tag for the release of asterisk-

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+Asterisk Bug Tracking Information
+To learn about and report Asterisk bugs, please visit 
+the official Asterisk Bug Tracker at:
+For more information on using the bug tracker, or to 
+learn how you can contribute by acting as a bug marshal
+please see:
+If you would like to submit a feature request, please
+resist the temptation to post it to the bug tracker.
+Feature requests should be posted to the asterisk-dev
+mailing list, located at:
+Thank you!
+=== 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 1.6.1 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. In addition, this behavior is 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 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.
+ * Added a new option "prematuremedia" that defaults to "yes". If you turn this
+   option on, chan_sip will not automatically initiate early media if it receives
+   audio from the incoming channel before there's been a progress indication.
+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.
+ * 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.
+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.
+ * 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.
+ * 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/ on LINUX
+   instead of the /var/run/ 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 '')
+ * 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.
+  * 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
+ * Added a new 'faxdetect=yes|no' configuration option to sip.conf.  When this
+    option is enabled, Asterisk will watch for a CNG tone in the incoming audio
+    for a received call.  If it is detected, the channel will jump to the 
+    'fax' extension in the dialplan.
+  * 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
+    is typically placed.
+  * The configuration method for the "joinempty" and "leavewhenempty" options has
+    changed to a comma-separated list of methods of determining member availability
+    instead of vague terms such as "yes," "loose," "no," and "strict." These old four
+    values are still accepted for backwards-compatibility, though.
+  * The average talktime is now calculated on queues. This information is reported via the
+    CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
+    and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
+    the queue.
+MeetMe Changes
+  * The 'o' option to provide an optimization has been removed and its functionality 
+     has been enabled by default.
+  * When a conference is created, the UNIQUEID of the channel that caused it to be
+     created is stored.  Then, every channel that joins the conference will have the
+     MEETMEUNIQUEID channel variable set with this ID.  This can be used to relate
+     callers that come and go from long standing conferences.
+  * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
+     except it does operations on a channel by name, instead of number in a conference.
+     This is a very useful feature in combination with the 'X' option to ChanSpy.
+  * Added 'C' option to Meetme which causes a caller to continue in the dialplan
+     when kicked out.
+  * Added new RealTime functionality to provide support for scheduled conferencing.
+     This includes optional messages to the caller if they attempt to join before
+     the schedule start time, or to allow the caller to join the conference early.
+     Also included is optional support for limiting the number of callers per
+     RealTime conference.
+  * Added the S() and L() options to the MeetMe application.  These are pretty
+     much identical to the S() and L() options to Dial().  They let you set
+     timeouts for the conference, as well as have warning sounds played to
+     let the caller know how much time is left, and when it is running out.
+  * Added the ability to do "meetme concise" with the "meetme" CLI command.
+     This extends the concise capabilities of this CLI command to include
+     listing all conferences, instead of an addition to the other sub commands
+     for the "meetme" command.