Update UPGRADE.txt file

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420607 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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 === UPGRADE-12.txt  -- Upgrade info for 11 to 12
 ===========================================================
 
+General Asterisk Changes:
+ - The asterisk command line -I option and the asterisk.conf internal_timing
+   option are removed and always enabled if any timing module is loaded.
+
+ - The per console verbose level feature as previously implemented caused a
+   large performance penalty.  The fix required some minor incompatibilities
+   if the new rasterisk is used to connect to an earlier version.  If the new
+   rasterisk connects to an older Asterisk version then the root console verbose
+   level is always affected by the "core set verbose" command of the remote
+   console even though it may appear to only affect the current console.  If
+   an older version of rasterisk connects to the new version then the
+   "core set verbose" command will have no effect.
+
+ - The asterisk compatibility options in asterisk.conf have been removed.
+   These options enabled certain backwards compatibility features for
+   pbx_realtime, res_agi, and app_set that made their behaviour similar to
+   Asterisk 1.4. Users who used these backwards compatibility settings should
+   update their dialplans to use ',' instead of '|' as a delimiter, and should
+   use the Set dialplan application instead of the MSet dialplan application.
+
+Build System:
+ - Sample config files have been moved from configs/ to a subfolder of that
+   directory, 'samples'.
+
+ - The menuselect utility has been pulled into the Asterisk repository. As a
+   result, the libxml2 development library is now a required dependency for
+   Asterisk.
+
+ - Added a new Compiler Flag, REF_DEBUG. When enabled, reference counted
+   objects will emit additional debug information to the refs log file located
+   in the standard Asterisk log file directory. This log file is useful in
+   tracking down object leaks and other reference counting issues. Prior to
+   this version, this option was only available by modifying the source code
+   directly. This change also includes a new script, refcounter.py, in the
+   contrib folder that will process the refs log file.
+
+Applications:
+
+ConfBridge:
+- The sound_place_into_conference sound used in Confbridge is now deprecated
+  and is no longer functional since it has been broken since its inception
+  and the fix involved using a different method to achieve the same goal. The
+  new method to achieve this functionality is by using sound_begin to play
+  a sound to the conference when waitmarked users are moved into the conference.
+
+SetMusicOnHold:
+ - The SetMusicOnHold dialplan application was deprecated and has been removed.
+   Users of the application should use the CHANNEL function's musicclass
+   setting instead.
+
+WaitMusicOnHold:
+ - The WaitMusicOnHold dialplan application was deprecated and has been
+   removed. Users of the application should use MusicOnHold with a duration
+   parameter instead.
+
+CDR Backends:
+ - The cdr_sqlite module was deprecated and has been removed. Users of this
+   module should use the cdr_sqlite3_custom module instead.
+
+Channel Drivers:
+
+chan_dahdi:
+ - SS7 support now requires libss7 v2.0 or later.
+
+ - Added the inband_on_setup_ack compatibility option to chan_dahdi.conf to
+   deal with switches that don't send an inband progress indication in the
+   SETUP ACKNOWLEDGE message.
+   Default is now no.
+
+chan_gtalk
+ - This module was deprecated and has been removed. Users of chan_gtalk
+   should use chan_motif.
+
+chan_h323
+ - This module was deprecated and has been removed. Users of chan_h323
+   should use chan_ooh323.
+
+chan_jingle
+ - This module was deprecated and has been removed. Users of chan_jingle
+   should use chan_motif.
+
+chan_pjsip:
+ - Added a 'force_avp' option to chan_pjsip which will force the usage of
+   'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' as the media transport type
+   in SDP offers depending on settings, even when DTLS is used for media
+   encryption.
+
+ - Added a 'media_use_received_transport' option to chan_pjsip which will
+   cause the SDP answer to use the media transport as received in the SDP
+   offer.
+
+chan_sip:
+ - Made set SIPREFERREDBYHDR as inheritable for better chan_pjsip
+   interoperability.
+
+ - The SIPPEER dialplan function no longer supports using a colon as a
+   delimiter for parameters. The parameters for the function should be
+   delimited using a comma.
+
+ - The SIPCHANINFO dialplan function was deprecated and has been removed. Users
+   of the function should use the CHANNEL function instead.
+
+ - Added a 'force_avp' option for chan_sip. When enabled this option will
+   cause the media transport in the offer or answer SDP to be 'RTP/AVP',
+   'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' even if a DTLS stream has been
+   configured. This option can be set to improve interoperability with WebRTC
+   clients that don't use the RFC defined transport for DTLS.
+
+ - The 'dtlsverify' option in chan_sip now has additional values besides
+   'yes' and 'no'. If 'yes' is specified both the certificate and fingerprint
+   will be verified. If 'no' is specified then neither the certificate or
+   fingerprint is verified. If 'certificate' is specified then only the
+   certificate is verified. If 'fingerprint' is specified then only the
+   fingerprint is verified.
+
+ - A 'dtlsfingerprint' option has been added to chan_sip which allows the
+   hash to be specified for the DTLS fingerprint placed in SDP. Supported
+   values are 'sha-1' and 'sha-256' with 'sha-256' being the default.
+
+ - The 'progressinband=never' option is now more zealous in the persecution of
+   progress messages coming from Asterisk. Channels bridged with a SIP channel
+   that has 'progressinband=never' set will not be able to forward their
+   progress indications through to the SIP device. chan_sip will now turn such
+   progress indications into a 180 Ringing (if a 180 has not yet been
+   transmitted) if 'progressinband=never'.
+
+  - The codec preference order in an SDP during an offer is slightly different
+    than previous releases. Prior to Asterisk 13, the preference order of
+    codecs used to be:
+    (1) Our preferred codec
+    (2) Our configured codecs
+    (3) Any non-audio joint codecs
+
+    One of the ways the new media format architecture in Asterisk 13 improves
+    performance is by reference counting formats, such that they can be reused
+    in many places without additional allocation. To not require a large
+    amount of locking, an instance of a format is immutable by convention.
+    This works well except for formats with attributes. Since a media format
+    with an attribute is a different object than the same format without an
+    attribute, we have to carry over the formats with attributes from an
+    inbound offer so that the correct attributes are offered in an outgoing
+    INVITE request. This requires some subtle tweaks to the preference order
+    to ensure that the media format with attributes is offered to a remote
+    peer, as opposed to the same media format (but without attributes) that
+    may be stored in the peer object.
+
+    All of this means that our offer offer list will now be:
+    (1) Our preferred codec
+    (2) Any joint codecs offered by the inbound offer
+    (3) All other codecs that are not the preferred codec and not a joint
+        codec offered by the inbound offer
+
+chan_unistim:
+ - The unistim.conf 'dateformat' has changed meaning of options values to conform
+   values used inside Unistim protocol
+
+ - Added 'dtmf_duration' option with changing default operation to disable
+   receivied dtmf playback on unistim phone
+
+Core:
+
+Account Codes:
+ - accountcode behavior changed somewhat to add functional peeraccount
+   support.  The main change is that local channels now cross accountcode
+   and peeraccount across the special bridge between the ;1 and ;2 channels
+   just like channels between normal bridges.  See the CHANGES file for
+   more information.
+
+ARI:
+ - The ARI version has been changed to 1.5.0. This is to reflect backwards
+   compatible changes made since 12.0.0 was released.
+
+ - Added a new ARI resource 'mailboxes' which allows the creation and
+   modification of mailboxes managed by external MWI. Modules res_mwi_external
+   and res_stasis_mailbox must be enabled to use this resource.
+
+ - Added new events for externally initiated transfers. The event
+   BridgeBlindTransfer is now raised when a channel initiates a blind transfer
+   of a bridge in the ARI controlled application to the dialplan; the
+   BridgeAttendedTransfer event is raised when a channel initiates an
+   attended transfer of a bridge in the ARI controlled application to the
+   dialplan.
+
+ - Channel variables may now be specified as a body parameter to the
+   POST /channels operation. The 'variables' key in the JSON is interpreted
+   as a sequence of key/value pairs that will be added to the created channel
+   as channel variables. Other parameters in the JSON body are treated as
+   query parameters of the same name.
+
+ - A bug fix in bridge creation has caused a behavioural change in how
+   subscriptions are created for bridges. A bridge created through ARI, does
+   not, by itself, have a subscription created for any particular Stasis
+   application. When a channel in a Stasis application joins a bridge, an
+   implicit event subscription is created for that bridge as well. Previously,
+   when a channel left such a bridge, the subscription was leaked; this allowed
+   for later bridge events to continue to be pushed to the subscribed
+   applications. That leak has been fixed; as a result, bridge events that were
+   delivered after a channel left the bridge are no longer delivered. An
+   application must subscribe to a bridge through the applications resource if
+   it wishes to receive all events related to a bridge.
+
+AMI:
+ - The AMI version has been changed to 2.5.0. This is to reflect backwards
+   compatible changes made since 12.0.0 was released.
+
+ - The DialStatus field in the DialEnd event can now have additional values.
+   This includes ABORT, CONTINUE, and GOTO.
+
+ - The res_mwi_external_ami module can, if loaded, provide additional AMI
+   actions and events that convey MWI state within Asterisk. This includes
+   the MWIGet, MWIUpdate, and MWIDelete actions, as well as the MWIGet and
+   MWIGetComplete events that occur in response to an MWIGet action.
+
+ - AMI now contains a new class authorization, 'security'. This is used with
+   the following new events: FailedACL, InvalidAccountID, SessionLimit,
+   MemoryLimit, LoadAverageLimit, RequestNotAllowed, AuthMethodNotAllowed,
+   RequestBadFormat, SuccessfulAuth, UnexpectedAddress, ChallengeResponseFailed,
+   InvalidPassword, ChallengeSent, and InvalidTransport.
+
+ - Bridge related events now have two additional fields: BridgeName and
+   BridgeCreator. BridgeName is a descriptive name for the bridge;
+   BridgeCreator is the name of the entity that created the bridge. This
+   affects the following events: ConfbridgeStart, ConfbridgeEnd,
+   ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
+   ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
+   AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
+
+ - MixMonitor AMI actions now require users to have authorization classes.
+   * MixMonitor - system
+   * MixMonitorMute - call or system
+   * StopMixMonitor - call or system
+
+ - Removed the undocumented manager.conf block-sockets option.  It interferes with
+   TCP/TLS inactivity timeouts.
+
+ - The response to the PresenceState AMI action has historically contained two
+   Message keys. The first of these is used as an informative message regarding
+   the success/failure of the action; the second contains a Presence state
+   specific message. Having two keys with the same unique name in an AMI
+   message is cumbersome for some client; hence, the Presence specific Message
+   has been deprecated. The message will now contain a PresenceMessage key
+   for the presence specific information; the Message key containing presence
+   information will be removed in the next major version of AMI.
+
+ - The manager.conf 'eventfilter' now takes an "extended" regular expression
+   instead of a "basic" one.
+
+CDRs:
+ - The "endbeforehexten" setting now defaults to "yes", instead of "no".
+   When set to "no", yhis setting will cause a new CDR to be generated when a
+   channel enters into hangup logic (either the 'h' extension or a hangup
+   handler subroutine). In general, this is not the preferred default: this
+   causes extra CDRs to be generated for a channel in many common dialplans.
+
+CLI commands:
+ - "core show settings" now lists the current console verbosity in addition
+   to the root console verbosity.
+
+ - "core set verbose" has not been able to support the by module verbose
+   logging levels since verbose logging levels were made per console.  That
+   syntax is now removed and a silence option added in its place.
+
+Logging:
+ - The 'verbose' setting in logger.conf still takes an optional argument,
+   specifying the verbosity level for each logging destination.  However,
+   the default is now to once again follow the current root console level.
+   As a result, using the AMI Command action with "core set verbose" could
+   again set the root console verbose level and affect the verbose level
+   logged.
+
+HTTP:
+ - Added http.conf session_inactivity timer option to close HTTP connections
+   that aren't doing anything.
+
+ - Added support for persistent HTTP connections.  To enable persistent
+   HTTP connections configure the keep alive time between HTTP requests.  The
+   keep alive time between HTTP requests is configured in http.conf with the
+   session_keep_alive parameter.
+
+Realtime Configuration:
+ - WARNING: The database migration script that adds the 'extensions' table for
+   realtime had to be modified due to an error when installing for MySQL.  The
+   'extensions' table's 'id' column was changed to be a primary key.  This could
+   potentially cause a migration problem.  If so, it may be necessary to
+   manually alter the affected table/column to bring it back in line with the
+   migration scripts.
+
+ - New columns have been added to realtime tables for 'support_path' on
+   ps_registrations and ps_aors and for 'path' on ps_contacts for the new
+   SIP Path support in chan_pjsip.
+
+ - The following new tables have been added for pjsip realtime: 'ps_systems',
+   'ps_globals', 'ps_tranports', 'ps_registrations'.
+
+ - The following columns were added to the 'ps_aors' realtime table:
+   'maximum_expiration', 'outbound_proxy', and 'support_path'.
+
+ - The following columns were added to the 'ps_contacts' realtime table:
+   'outbound_proxy', 'user_agent', and 'path'.
+
+ - New columns have been added to the ps_endpoints realtime table for the
+   'media_address', 'redirect_method' and 'set_var' options.  Also the
+   'mwi_fromuser' column was renamed to 'mwi_from_user'. A new column
+   'message_context' was added to let users configure how MESSAGE requests are
+   routed to the dialplan.
+
+ - A new column was added to the 'ps_globals' realtime table for the 'debug'
+   option.
+
+ - PJSIP endpoint columns 'tos_audio' and 'tos_video' have been changed from
+   yes/no enumerators to string values. 'cos_audio' and 'cos_video' have been
+   changed from yes/no enumerators to integer values. PJSIP transport column
+   'tos' has been changed from a yes/no enumerator to a string value. 'cos' has
+   been changed from a yes/no enumerator to an integer value.
+
+ - The 'queues' and 'queue_members' realtime tables have been added to the
+   config Alembic scripts.
+
+ - A new set of Alembic scripts has been added for CDR tables. This will create
+   a 'cdr' table with the default schema that Asterisk expects.
+
+Resources:
+
+res_odbc:
+- The compatibility setting, allow_empty_string_in_nontext, has been removed.
+  Empty column values will be stored as empty strings during realtime updates.
+
+res_jabber:
+ - This module was deprecated and has been removed. Users of this module should
+   use res_xmpp instead.
+
+res_http_websocket:
+ - Added a compatibility option to ari.conf, sip.conf, and pjsip.conf
+   'websocket_write_timeout'. When a websocket connection exists where Asterisk
+   writes a substantial amount of data to the connected client, and the connected
+   client is slow to process the received data, the socket may be disconnected.
+   In such cases, it may be necessary to adjust this value.
+   Default is 100 ms.
+
+Scripts:
+
+safe_asterisk:
+ - The safe_asterisk script was previously not installed on top of an existing
+   version. This caused bug-fixes in that script not to be deployed. If your
+   safe_asterisk script is customized, be sure to keep your changes. Custom
+   values for variables should be created in *.sh file(s) inside
+   ASTETCDIR/startup.d/. See ASTERISK-21965.
+
+ - Changed a log message in safe_asterisk and the $NOTIFY mail subject. If
+   you use tools to parse either of them, update your parse functions
+   accordingly. The changed strings are:
+   - "Exited on signal $EXITSIGNAL" => "Asterisk exited on signal $EXITSIGNAL."
+   - "Asterisk Died" => "Asterisk on $MACHINE died (sig $EXITSIGNAL)"
+
+Utilities:
+ - The refcounter program has been removed in favor of the refcounter.py script
+   in contrib/scripts.
 
 ===========================================================
 ===========================================================