From 9d79ccd3db8c7caf25aa46981a15c2b7bf8e0bfd Mon Sep 17 00:00:00 2001 From: Matthew Jordan Date: Sat, 11 Aug 2012 19:13:55 +0000 Subject: 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 --- UPGRADE-11.txt | 226 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 226 insertions(+) create mode 100644 UPGRADE-11.txt (limited to 'UPGRADE-11.txt') diff --git a/UPGRADE-11.txt b/UPGRADE-11.txt new file mode 100644 index 000000000..d565602d9 --- /dev/null +++ b/UPGRADE-11.txt @@ -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 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,) 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. + +=========================================================== +=========================================================== -- cgit v1.2.3