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	Add UPGRADE-11.txt file; update UPGRADE.txt to reflect Asterisk 12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371170 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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							| @@ -0,0 +1,226 @@ | ||||
| =========================================================== | ||||
| === | ||||
| === 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 | ||||
| === | ||||
| =========================================================== | ||||
|  | ||||
| From 10 to 11: | ||||
|  | ||||
| Voicemail: | ||||
|  - All voicemails now have a "msg_id" which uniquely identifies a message. For | ||||
|    users of filesystem and IMAP storage of voicemail, this should be transparent. | ||||
|    For users of ODBC, you will need to add a "msg_id" column to your voice mail | ||||
|    messages table. This should be a string capable of holding at least 32 characters. | ||||
|    All messages created in old Asterisk installations will have a msg_id added to | ||||
|    them when required. This operation should be transparent as well. | ||||
|  | ||||
| Parking: | ||||
|  - The comebacktoorigin setting must now be set per parking lot. The setting in | ||||
|    the general section will not be applied automatically to each parking lot. | ||||
|  - The BLINDTRANSFER channel variable is deleted from a channel when it is | ||||
|    bridged to prevent subtle bugs in the parking feature.  The channel | ||||
|    variable is used by Asterisk internally for the Park application to work | ||||
|    properly.  If you were using it for your own purposes, copy it to your | ||||
|    own channel variable before the channel is bridged. | ||||
|  | ||||
| res_ais: | ||||
|  - Users of res_ais in versions of Asterisk prior to Asterisk 11 must change | ||||
|    to use the res_corosync module, instead.  OpenAIS is deprecated, but | ||||
|    Corosync is still actively developed and maintained.  Corosync came out of | ||||
|    the OpenAIS project. | ||||
|  | ||||
| Dialplan Functions: | ||||
|  - MAILBOX_EXISTS has been deprecated. Use VM_INFO with the 'exists' parameter | ||||
|    instead. | ||||
|  - Macro has been deprecated in favor of GoSub.  For redirecting and connected | ||||
|    line purposes use the following variables instead of their macro equivalents: | ||||
|    REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, | ||||
|    CONNECTED_LINE_SEND_SUB, CONNECTED_LINE_SEND_SUB_ARGS. | ||||
|  - The REDIRECTING function now supports the redirecting original party id | ||||
|    and reason. | ||||
|  - The HANGUPCAUSE and HANGUPCAUSE_KEYS functions have been introduced to | ||||
|    provide a replacement for the SIP_CAUSE hash. The HangupCauseClear | ||||
|    application has also been introduced to remove this data from the channel | ||||
|    when necessary. | ||||
|  | ||||
|  | ||||
| func_enum: | ||||
|  - ENUM query functions now return a count of -1 on lookup error to | ||||
|    differentiate between a failed query and a successful query with 0 results | ||||
|    matching the specified type. | ||||
|  | ||||
| CDR: | ||||
|  - cdr_adaptive_odbc now supports specifying a schema so that Asterisk can | ||||
|    connect to databases that use schemas. | ||||
|  | ||||
| Configuration Files: | ||||
|  - Files listed below have been updated to be more consistent with how Asterisk | ||||
|    parses configuration files.  This makes configuration files more consistent | ||||
|    with what is expected across modules. | ||||
|  | ||||
|    - cdr.conf: [general] and [csv] sections | ||||
|    - dnsmgr.conf | ||||
|    - dsp.conf | ||||
|  | ||||
|  - The 'verbose' setting in logger.conf now takes an optional argument, | ||||
|    specifying the verbosity level for each logging destination.  The default, | ||||
|    if not otherwise specified, is a verbosity of 3. | ||||
|  | ||||
| AMI: | ||||
|   - DBDelTree now correctly returns an error when 0 rows are deleted just as | ||||
|     the DBDel action does. | ||||
|   - The IAX2 PeerStatus event now sends a 'Port' header.  In Asterisk 10, this was | ||||
|     erroneously being sent as a 'Post' header. | ||||
|  | ||||
| CCSS: | ||||
|  - Macro is deprecated. Use cc_callback_sub instead of cc_callback_macro | ||||
|    in channel configurations. | ||||
|  | ||||
| app_meetme: | ||||
|   - The 'c' option (announce user count) will now work even if the 'q' (quiet) | ||||
|     option is enabled. | ||||
|  | ||||
| app_followme: | ||||
|  - Answered outgoing calls no longer get cut off when the next step is started. | ||||
|    You now have until the last step times out to decide if you want to accept | ||||
|    the call or not before being disconnected. | ||||
|  | ||||
| chan_gtalk: | ||||
|  - chan_gtalk has been deprecated in favor of the chan_motif channel driver. It is recommended | ||||
|    that users switch to using it as it is a core supported module. | ||||
|  | ||||
| chan_jingle: | ||||
|  - chan_jingle has been deprecated in favor of the chan_motif channel driver. It is recommended | ||||
|    that users switch to using it as it is a core supported module. | ||||
|  | ||||
| SIP | ||||
| === | ||||
|  - A new option "tonezone" for setting default tonezone for the channel driver | ||||
|    or individual devices | ||||
|  - A new manager event, "SessionTimeout" has been added and is triggered when | ||||
|    a call is terminated due to RTP stream inactivity or SIP session timer | ||||
|    expiration. | ||||
|  - SIP_CAUSE is now deprecated.  It has been modified to use the same | ||||
|    mechanism as the HANGUPCAUSE function.  Behavior should not change, but | ||||
|    performance should be vastly improved.  The HANGUPCAUSE function should now | ||||
|    be used instead of SIP_CAUSE. Because of this, the storesipcause option in | ||||
|    sip.conf is also deprecated. | ||||
|  - The sip paramater for Originating Line Information (oli, isup-oli, and | ||||
|    ss7-oli) is now parsed out of the From header and copied into the channel's | ||||
|    ANI2 information field.  This is readable from the CALLERID(ani2) dialplan | ||||
|    function. | ||||
|  - ICE support has been added and is enabled by default. Some endpoints may have | ||||
|    problems with the ICE candidates within the SDP. If this is the case ICE support | ||||
|    can be disabled globally or on a per-endpoint basis using the icesupport | ||||
|    configuration option. Symptoms of this include one way media or no media flow. | ||||
|  | ||||
| chan_unistim | ||||
|  - Due to massive update in chan_unistim phone keys functions and on-screen  | ||||
|    information changed. | ||||
|  | ||||
| users.conf: | ||||
|  - A defined user with hasvoicemail=yes now finally uses a Gosub to stdexten | ||||
|    as documented in extensions.conf.sample since v1.6.0 instead of a Macro as | ||||
|    documented in v1.4.  Set the asterisk.conf stdexten=macro parameter to | ||||
|    invoke the stdexten the old way. | ||||
|  | ||||
| res_jabber | ||||
|  - This module has been deprecated in favor of the res_xmpp module. The res_xmpp | ||||
|    module is backwards compatible with the res_jabber configuration file, dialplan | ||||
|    functions, and AMI actions. The old CLI commands can also be made available using | ||||
|    the res_clialiases template for Asterisk 11. | ||||
|  | ||||
| From 1.8 to 10: | ||||
|  | ||||
| cel_pgsql: | ||||
|  - This module now expects an 'extra' column in the database for data added | ||||
|    using the CELGenUserEvent() application. | ||||
|  | ||||
| ConfBridge | ||||
|  - ConfBridge's dialplan arguments have changed and are not | ||||
|    backwards compatible. | ||||
|  | ||||
| File Interpreters | ||||
|  - The format interpreter formats/format_sln16.c for the file extension | ||||
|    '.sln16' has been removed. The '.sln16' file interpreter now exists | ||||
|    in the formats/format_sln.c module along with new support for sln12, | ||||
|    sln24, sln32, sln44, sln48, sln96, and sln192 file extensions. | ||||
|  | ||||
| HTTP: | ||||
|  - A bindaddr must be specified in order for the HTTP server | ||||
|    to run. Previous versions would default to 0.0.0.0 if no | ||||
|    bindaddr was specified. | ||||
|  | ||||
| Gtalk: | ||||
|  - The default value for 'context' and 'parkinglots' in gtalk.conf has | ||||
|    been changed to 'default', previously they were empty. | ||||
|  | ||||
| chan_dahdi: | ||||
|  - The mohinterpret=passthrough setting is deprecated in favor of | ||||
|    moh_signaling=notify. | ||||
|  | ||||
| pbx_lua: | ||||
|  - Execution no longer continues after applications that do dialplan jumps | ||||
|    (such as app.goto).  Now when an application such as app.goto() is called, | ||||
|    control is returned back to the pbx engine and the current extension | ||||
|    function stops executing. | ||||
|  - the autoservice now defaults to being on by default | ||||
|  - autoservice_start() and autoservice_start() no longer return a value. | ||||
|  | ||||
| Queue: | ||||
|  - Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members | ||||
|  - QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty. | ||||
|  | ||||
| Asterisk Database: | ||||
|  - The internal Asterisk database has been switched from Berkeley DB 1.86 to | ||||
|    SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3 | ||||
|    utility in the UTILS section of menuselect. If an existing astdb is found and no | ||||
|    astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will | ||||
|    convert an existing astdb to the SQLite3 version automatically at runtime. If | ||||
|    moving back from Asterisk 10 to Asterisk 1.8, the astdb2bdb utility can be used | ||||
|    to create a Berkeley DB copy of the SQLite3 astdb that Asterisk 10 uses. | ||||
|  | ||||
| Manager: | ||||
|  - The AMI protocol version was incremented to 1.2 as a result of changing two | ||||
|    instances of the Unlink event to Bridge events. This change was documented | ||||
|    as part of the AMI 1.1 update, but two Unlink events were inadvertently left | ||||
|    unchanged. | ||||
|  | ||||
| Module Support Level | ||||
|  - All modules in the addons, apps, bridge, cdr, cel, channels, codecs,  | ||||
|    formats, funcs, pbx, and res have been updated to include MODULEINFO data | ||||
|    that includes <support_level> tags with a value of core, extended, or deprecated. | ||||
|    More information is available on the Asterisk wiki at  | ||||
|    https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States | ||||
|  | ||||
|    Deprecated modules are now marked to not build by default and must be explicitly | ||||
|    enabled in menuselect. | ||||
|  | ||||
| chan_sip: | ||||
|  - Setting of HASH(SIP_CAUSE,<slave-channel-name>) on channels is now disabled | ||||
|    by default. It can be enabled using the 'storesipcause' option. This feature | ||||
|    has a significant performance penalty. | ||||
|  | ||||
| UDPTL: | ||||
|  - The default UDPTL port range in udptl.conf.sample differed from the defaults | ||||
|    in the source. If you didn't have a config file, you got 4500 to 4599. Now the | ||||
|    default is 4000 to 4999. | ||||
|  | ||||
| =========================================================== | ||||
| =========================================================== | ||||
| @@ -17,9 +17,14 @@ | ||||
| === 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 | ||||
| === | ||||
| =========================================================== | ||||
|  | ||||
| From 11 to 12: | ||||
|  | ||||
|  | ||||
|  | ||||
| From 10 to 11: | ||||
|  | ||||
| Voicemail: | ||||
|   | ||||
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