asterisk/UPGRADE.txt

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===
=== Information for upgrading between Asterisk versions
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also include advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
=== UPGRADE-10.txt  -- Upgrade info for 1.8 to 10
=== UPGRADE-11.txt  -- Upgrade info for 10 to 11
=== UPGRADE-12.txt  -- Upgrade info for 11 to 12
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From 12 to 13:

 - 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.

 - 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.

 - 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.

 - 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.

ARI:
 - The ARI version has been changed from 1.0.0 to 1.1.0. This is to reflect
   the backwards compatible changes listed below.

 - 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 from 2.0.0 to 2.1.0. This is to reflect
   the backwards compatible changes listed below.

 - 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.

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.

 - The cdr_sqlite module was deprecated and has been removed. Users of this
   module should use the cdr_sqlite3_custom module instead.

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

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.

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.


Configuration Files:
 - 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.

 - The manager.conf 'eventfilter' now takes an "extended" regular expression
   instead of a "basic" one.

 - The unistim.conf 'dateformat' has changed meaning of options values to conform
   values used inside Unistim protocol

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.

MusicOnHold
 - The SetMusicOnHold dialplan application was deprecated and has been removed.
   Users of the application should use the CHANNEL function's musicclass
   setting instead.

 - The WaitMusicOnHold dialplan application was deprecated and has been
   removed. Users of the application should use MusicOnHold with a duration
   parameter instead.

ODBC:
- The compatibility setting, allow_empty_string_in_nontext, has been removed.
  Empty column values will be stored as empty strings during realtime updates.

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.

res_jabber:
 - This module was deprecated and has been removed. Users of this module should
   use res_xmpp instead.

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)"

Unistim:
 - Added 'dtmf_duration' option with changing default operation to disable
   receivied dtmf playback on unistim phone

Utilities:
 - The refcounter program has been removed in favor of the refcounter.py script
   in contrib/scripts.

WebSockets:
 - 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.

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