........ Merged revisions 414345 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414346 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414347 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
202 lines
9.8 KiB
Text
202 lines
9.8 KiB
Text
===========================================================
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===
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=== Information for upgrading between Asterisk versions
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===
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=== These files document all the changes that MUST be taken
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=== into account when upgrading between the Asterisk
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=== versions listed below. These changes may require that
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=== you modify your configuration files, dialplan or (in
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=== some cases) source code if you have your own Asterisk
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=== modules or patches. These files also include advance
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=== notice of any functionality that has been marked as
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=== 'deprecated' and may be removed in a future release,
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=== along with the suggested replacement functionality.
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===
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=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
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=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
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=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
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=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
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=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
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=== UPGRADE-11.txt -- Upgrade info for 10 to 11
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=== UPGRADE-12.txt -- Upgrade info for 11 to 12
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===========================================================
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From 12 to 13:
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- The asterisk command line -I option and the asterisk.conf internal_timing
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option are removed and always enabled if any timing module is loaded.
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- The per console verbose level feature as previously implemented caused a
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large performance penalty. The fix required some minor incompatibilities
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if the new rasterisk is used to connect to an earlier version. If the new
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rasterisk connects to an older Asterisk version then the root console verbose
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level is always affected by the "core set verbose" command of the remote
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console even though it may appear to only affect the current console. If
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an older version of rasterisk connects to the new version then the
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"core set verbose" command will have no effect.
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- Added a new Compiler Flag, REF_DEBUG. When enabled, reference counted
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objects will emit additional debug information to the refs log file located
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in the standard Asterisk log file directory. This log file is useful in
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tracking down object leaks and other reference counting issues. Prior to
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this version, this option was only available by modifying the source code
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directly. This change also includes a new script, refcounter.py, in the
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contrib folder that will process the refs log file.
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ARI:
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- The ARI version has been changed from 1.0.0 to 1.1.0. This is to reflect
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the backwards compatible changes listed below.
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- Added a new ARI resource 'mailboxes' which allows the creation and
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modification of mailboxes managed by external MWI. Modules res_mwi_external
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and res_stasis_mailbox must be enabled to use this resource.
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- Added new events for externally initiated transfers. The event
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BridgeBlindTransfer is now raised when a channel initiates a blind transfer
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of a bridge in the ARI controlled application to the dialplan; the
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BridgeAttendedTransfer event is raised when a channel initiates an
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attended transfer of a bridge in the ARI controlled application to the
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dialplan.
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- Channel variables may now be specified as a body parameter to the
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POST /channels operation. The 'variables' key in the JSON is interpreted
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as a sequence of key/value pairs that will be added to the created channel
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as channel variables. Other parameters in the JSON body are treated as
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query parameters of the same name.
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- A bug fix in bridge creation has caused a behavioural change in how
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subscriptions are created for bridges. A bridge created through ARI, does
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not, by itself, have a subscription created for any particular Stasis
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application. When a channel in a Stasis application joins a bridge, an
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implicit event subscription is created for that bridge as well. Previously,
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when a channel left such a bridge, the subscription was leaked; this allowed
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for later bridge events to continue to be pushed to the subscribed
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applications. That leak has been fixed; as a result, bridge events that were
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delivered after a channel left the bridge are no longer delivered. An
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application must subscribe to a bridge through the applications resource if
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it wishes to receive all events related to a bridge.
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AMI:
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- The AMI version has been changed from 2.0.0 to 2.1.0. This is to reflect
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the backwards compatible changes listed below.
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- The DialStatus field in the DialEnd event can now have additional values.
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This includes ABORT, CONTINUE, and GOTO.
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- The res_mwi_external_ami module can, if loaded, provide additional AMI
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actions and events that convey MWI state within Asterisk. This includes
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the MWIGet, MWIUpdate, and MWIDelete actions, as well as the MWIGet and
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MWIGetComplete events that occur in response to an MWIGet action.
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- AMI now contains a new class authorization, 'security'. This is used with
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the following new events: FailedACL, InvalidAccountID, SessionLimit,
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MemoryLimit, LoadAverageLimit, RequestNotAllowed, AuthMethodNotAllowed,
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RequestBadFormat, SuccessfulAuth, UnexpectedAddress, ChallengeResponseFailed,
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InvalidPassword, ChallengeSent, and InvalidTransport.
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- Bridge related events now have two additional fields: BridgeName and
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BridgeCreator. BridgeName is a descriptive name for the bridge;
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BridgeCreator is the name of the entity that created the bridge. This
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affects the following events: ConfbridgeStart, ConfbridgeEnd,
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ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
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ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
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AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
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CDRs:
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- The "endbeforehexten" setting now defaults to "yes", instead of "no".
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When set to "no", yhis setting will cause a new CDR to be generated when a
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channel enters into hangup logic (either the 'h' extension or a hangup
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handler subroutine). In general, this is not the preferred default: this
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causes extra CDRs to be generated for a channel in many common dialplans.
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chan_sip:
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- Made set SIPREFERREDBYHDR as inheritable for better chan_pjsip
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interoperability.
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CLI commands:
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- "core show settings" now lists the current console verbosity in addition
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to the root console verbosity.
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- "core set verbose" has not been able to support the by module verbose
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logging levels since verbose logging levels were made per console. That
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syntax is now removed and a silence option added in its place.
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ConfBridge:
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- The sound_place_into_conference sound used in Confbridge is now deprecated
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and is no longer functional since it has been broken since its inception
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and the fix involved using a different method to achieve the same goal. The
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new method to achieve this functionality is by using sound_begin to play
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a sound to the conference when waitmarked users are moved into the conference.
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Configuration Files:
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- The 'verbose' setting in logger.conf still takes an optional argument,
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specifying the verbosity level for each logging destination. However,
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the default is now to once again follow the current root console level.
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As a result, using the AMI Command action with "core set verbose" could
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again set the root console verbose level and affect the verbose level
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logged.
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- The manager.conf 'eventfilter' now takes an "extended" regular expression
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instead of a "basic" one.
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- The unistim.conf 'dateformat' has changed meaning of options values to conform
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values used inside Unistim protocol
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ODBC:
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- The compatibility setting, allow_empty_string_in_nontext, has been removed.
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Empty column values will be stored as empty strings during realtime updates.
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Realtime Configuration:
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- WARNING: The database migration script that adds the 'extensions' table for
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realtime had to be modified due to an error when installing for MySQL. The
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'extensions' table's 'id' column was changed to be a primary key. This could
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potentially cause a migration problem. If so, it may be necessary to
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manually alter the affected table/column to bring it back in line with the
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migration scripts.
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- New columns have been added to realtime tables for 'support_path' on
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ps_registrations and ps_aors and for 'path' on ps_contacts for the new
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SIP Path support in chan_pjsip.
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- The following new tables have been added for pjsip realtime: 'ps_systems',
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'ps_globals', 'ps_tranports', 'ps_registrations'.
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- The following columns were added to the 'ps_aors' realtime table:
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'maximum_expiration', 'outbound_proxy', and 'support_path'.
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- The following columns were added to the 'ps_contacts' realtime table:
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'outbound_proxy', 'user_agent', and 'path'.
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- New columns have been added to the ps_endpoints realtime table for the
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'media_address', 'redirect_method' and 'set_var' options. Also the
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'mwi_fromuser' column was renamed to 'mwi_from_user'. A new column
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'message_context' was added to let users configure how MESSAGE requests are
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routed to the dialplan.
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- A new column was added to the 'ps_globals' realtime table for the 'debug'
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option.
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- PJSIP endpoint columns 'tos_audio' and 'tos_video' have been changed from
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yes/no enumerators to string values. 'cos_audio' and 'cos_video' have been
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changed from yes/no enumerators to integer values. PJSIP transport column
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'tos' has been changed from a yes/no enumerator to a string value. 'cos' has
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been changed from a yes/no enumerator to an integer value.
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- The 'queues' and 'queue_members' realtime tables have been added to the
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config Alembic scripts.
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- A new set of Alembic scripts has been added for CDR tables. This will create
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a 'cdr' table with the default schema that Asterisk expects.
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Unistim:
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- Added 'dtmf_duration' option with changing default operation to disable
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receivied dtmf playback on unistim phone
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Utilities:
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- The refcounter program has been removed in favor of the refcounter.py script
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in contrib/scripts.
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===========================================================
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===========================================================
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