[svn-commits] mjordan: trunk r371170 - in /trunk: UPGRADE-11.txt UPGRADE.txt
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Sat Aug 11 14:14:00 CDT 2012
Author: mjordan
Date: Sat Aug 11 14:13:55 2012
New Revision: 371170
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=371170
Log:
Add UPGRADE-11.txt file; update UPGRADE.txt to reflect Asterisk 12
Added:
trunk/UPGRADE-11.txt (with props)
Modified:
trunk/UPGRADE.txt
Added: trunk/UPGRADE-11.txt
URL: http://svnview.digium.com/svn/asterisk/trunk/UPGRADE-11.txt?view=auto&rev=371170
==============================================================================
--- trunk/UPGRADE-11.txt (added)
+++ trunk/UPGRADE-11.txt Sat Aug 11 14:13:55 2012
@@ -1,0 +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.
+
+===========================================================
+===========================================================
Propchange: trunk/UPGRADE-11.txt
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Propchange: trunk/UPGRADE-11.txt
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svn:keywords = Author Date Id Revision
Propchange: trunk/UPGRADE-11.txt
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svn:mime-type = text/plain
Modified: trunk/UPGRADE.txt
URL: http://svnview.digium.com/svn/asterisk/trunk/UPGRADE.txt?view=diff&rev=371170&r1=371169&r2=371170
==============================================================================
--- trunk/UPGRADE.txt (original)
+++ trunk/UPGRADE.txt Sat Aug 11 14:13:55 2012
@@ -17,8 +17,13 @@
=== 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:
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