(issue AST-969) Reported by John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
239 lines
10 KiB
Text
239 lines
10 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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===
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===========================================================
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From 11 to 12:
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AMI:
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- The SIP SIPqualifypeer action now sends a response indicating it will qualify
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a peer once a peer has been found to qualify. Once the qualify has been
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completed it will now issue a SIPqualifypeerdone event.
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Queue:
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- Queue logging for PAUSEALL/UNPAUSEALL now only occurs if the interface this is
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performed on is a member of at least one queue.
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From 10 to 11:
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Voicemail:
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- All voicemails now have a "msg_id" which uniquely identifies a message. For
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users of filesystem and IMAP storage of voicemail, this should be transparent.
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For users of ODBC, you will need to add a "msg_id" column to your voice mail
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messages table. This should be a string capable of holding at least 32 characters.
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All messages created in old Asterisk installations will have a msg_id added to
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them when required. This operation should be transparent as well.
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Parking:
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- The comebacktoorigin setting must now be set per parking lot. The setting in
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the general section will not be applied automatically to each parking lot.
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- The BLINDTRANSFER channel variable is deleted from a channel when it is
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bridged to prevent subtle bugs in the parking feature. The channel
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variable is used by Asterisk internally for the Park application to work
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properly. If you were using it for your own purposes, copy it to your
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own channel variable before the channel is bridged.
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res_ais:
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- Users of res_ais in versions of Asterisk prior to Asterisk 11 must change
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to use the res_corosync module, instead. OpenAIS is deprecated, but
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Corosync is still actively developed and maintained. Corosync came out of
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the OpenAIS project.
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Dialplan Functions:
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- MAILBOX_EXISTS has been deprecated. Use VM_INFO with the 'exists' parameter
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instead.
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- Macro has been deprecated in favor of GoSub. For redirecting and connected
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line purposes use the following variables instead of their macro equivalents:
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REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS,
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CONNECTED_LINE_SEND_SUB, CONNECTED_LINE_SEND_SUB_ARGS.
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- The REDIRECTING function now supports the redirecting original party id
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and reason.
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- The HANGUPCAUSE and HANGUPCAUSE_KEYS functions have been introduced to
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provide a replacement for the SIP_CAUSE hash. The HangupCauseClear
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application has also been introduced to remove this data from the channel
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when necessary.
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func_enum:
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- ENUM query functions now return a count of -1 on lookup error to
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differentiate between a failed query and a successful query with 0 results
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matching the specified type.
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CDR:
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- cdr_adaptive_odbc now supports specifying a schema so that Asterisk can
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connect to databases that use schemas.
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Configuration Files:
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- Files listed below have been updated to be more consistent with how Asterisk
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parses configuration files. This makes configuration files more consistent
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with what is expected across modules.
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- cdr.conf: [general] and [csv] sections
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- dnsmgr.conf
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- dsp.conf
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- The 'verbose' setting in logger.conf now takes an optional argument,
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specifying the verbosity level for each logging destination. The default,
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if not otherwise specified, is a verbosity of 3.
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AMI:
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- DBDelTree now correctly returns an error when 0 rows are deleted just as
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the DBDel action does.
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- The IAX2 PeerStatus event now sends a 'Port' header. In Asterisk 10, this was
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erroneously being sent as a 'Post' header.
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CCSS:
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- Macro is deprecated. Use cc_callback_sub instead of cc_callback_macro
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in channel configurations.
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app_meetme:
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- The 'c' option (announce user count) will now work even if the 'q' (quiet)
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option is enabled.
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app_followme:
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- Answered outgoing calls no longer get cut off when the next step is started.
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You now have until the last step times out to decide if you want to accept
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the call or not before being disconnected.
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chan_gtalk:
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- chan_gtalk has been deprecated in favor of the chan_motif channel driver. It is recommended
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that users switch to using it as it is a core supported module.
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chan_jingle:
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- chan_jingle has been deprecated in favor of the chan_motif channel driver. It is recommended
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that users switch to using it as it is a core supported module.
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SIP
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===
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- A new option "tonezone" for setting default tonezone for the channel driver
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or individual devices
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- A new manager event, "SessionTimeout" has been added and is triggered when
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a call is terminated due to RTP stream inactivity or SIP session timer
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expiration.
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- SIP_CAUSE is now deprecated. It has been modified to use the same
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mechanism as the HANGUPCAUSE function. Behavior should not change, but
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performance should be vastly improved. The HANGUPCAUSE function should now
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be used instead of SIP_CAUSE. Because of this, the storesipcause option in
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sip.conf is also deprecated.
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- The sip paramater for Originating Line Information (oli, isup-oli, and
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ss7-oli) is now parsed out of the From header and copied into the channel's
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ANI2 information field. This is readable from the CALLERID(ani2) dialplan
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function.
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- ICE support has been added and is enabled by default. Some endpoints may have
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problems with the ICE candidates within the SDP. If this is the case ICE support
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can be disabled globally or on a per-endpoint basis using the icesupport
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configuration option. Symptoms of this include one way media or no media flow.
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chan_unistim
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- Due to massive update in chan_unistim phone keys functions and on-screen
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information changed.
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users.conf:
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- A defined user with hasvoicemail=yes now finally uses a Gosub to stdexten
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as documented in extensions.conf.sample since v1.6.0 instead of a Macro as
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documented in v1.4. Set the asterisk.conf stdexten=macro parameter to
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invoke the stdexten the old way.
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res_jabber
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- This module has been deprecated in favor of the res_xmpp module. The res_xmpp
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module is backwards compatible with the res_jabber configuration file, dialplan
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functions, and AMI actions. The old CLI commands can also be made available using
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the res_clialiases template for Asterisk 11.
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From 1.8 to 10:
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cel_pgsql:
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- This module now expects an 'extra' column in the database for data added
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using the CELGenUserEvent() application.
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ConfBridge
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- ConfBridge's dialplan arguments have changed and are not
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backwards compatible.
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File Interpreters
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- The format interpreter formats/format_sln16.c for the file extension
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'.sln16' has been removed. The '.sln16' file interpreter now exists
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in the formats/format_sln.c module along with new support for sln12,
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sln24, sln32, sln44, sln48, sln96, and sln192 file extensions.
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HTTP:
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- A bindaddr must be specified in order for the HTTP server
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to run. Previous versions would default to 0.0.0.0 if no
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bindaddr was specified.
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Gtalk:
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- The default value for 'context' and 'parkinglots' in gtalk.conf has
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been changed to 'default', previously they were empty.
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chan_dahdi:
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- The mohinterpret=passthrough setting is deprecated in favor of
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moh_signaling=notify.
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pbx_lua:
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- Execution no longer continues after applications that do dialplan jumps
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(such as app.goto). Now when an application such as app.goto() is called,
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control is returned back to the pbx engine and the current extension
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function stops executing.
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- the autoservice now defaults to being on by default
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- autoservice_start() and autoservice_start() no longer return a value.
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Queue:
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- Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
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- QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.
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Asterisk Database:
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- The internal Asterisk database has been switched from Berkeley DB 1.86 to
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SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
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utility in the UTILS section of menuselect. If an existing astdb is found and no
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astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
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convert an existing astdb to the SQLite3 version automatically at runtime. If
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moving back from Asterisk 10 to Asterisk 1.8, the astdb2bdb utility can be used
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to create a Berkeley DB copy of the SQLite3 astdb that Asterisk 10 uses.
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Manager:
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- The AMI protocol version was incremented to 1.2 as a result of changing two
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instances of the Unlink event to Bridge events. This change was documented
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as part of the AMI 1.1 update, but two Unlink events were inadvertently left
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unchanged.
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Module Support Level
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- All modules in the addons, apps, bridge, cdr, cel, channels, codecs,
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formats, funcs, pbx, and res have been updated to include MODULEINFO data
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that includes <support_level> tags with a value of core, extended, or deprecated.
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More information is available on the Asterisk wiki at
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https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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Deprecated modules are now marked to not build by default and must be explicitly
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enabled in menuselect.
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chan_sip:
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- Setting of HASH(SIP_CAUSE,<slave-channel-name>) on channels is now disabled
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by default. It can be enabled using the 'storesipcause' option. This feature
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has a significant performance penalty.
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UDPTL:
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- The default UDPTL port range in udptl.conf.sample differed from the defaults
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in the source. If you didn't have a config file, you got 4500 to 4599. Now the
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default is 4000 to 4999.
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===========================================================
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===========================================================
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