=========================================================== === === 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 === UPGRADE-11.txt -- Upgrade info for 10 to 11 === =========================================================== AMI: - The SIP SIPqualifypeer action now sends a response indicating it will qualify a peer once a peer has been found to qualify. Once the qualify has been completed it will now issue a SIPqualifypeerdone event. - Version 1.4 - The details of what happens to a channel when a masquerade happens (transfers, parking, etc) have changed. - The Masquerade event now includes the Uniqueid's of the clone and original channels. - Channels no longer swap Uniqueid's as a result of the masquerade. - Instead of a shell game of renames, there's now a single rename, appending to the name of the original channel. CEL: - The Uniqueid field for a channel is now a stable identifier, and will not change due to transfers, parking, etc. Queues: - Queue logging for PAUSEALL/UNPAUSEALL now only occurs if the interface this is performed on is a member of at least one queue. - Queue strategy rrmemory now has a predictable order similar to strategy rrordered. Members will be called in the order that they are added to the queue. - CDR behavior in app_queue has been modified slightly. The CDR record will now only record a disposition of BUSY if all Queue members were actually busy on a call or some Queue members were busy or paused. Previously, any Queue member being paused would result in a disposition of BUSY. - Removed the queues.conf check_state_unknown option. It is no longer necessary. Dial: - Now recognizes 'W' to pause sending DTMF for one second in addition to the previously existing 'w' that paused sending DTMF for half a second. ExternalIVR: - Now recognizes 'W' to pause sending DTMF for one second in addition to the previously existing 'w' that paused sending DTMF for half a second. SendDTMF: - Now recognizes 'W' to pause sending DTMF for one second in addition to the previously existing 'w' that paused sending DTMF for half a second. chan_dahdi: - Analog port dialing and deferred DTMF dialing for PRI now distinguishes between 'w' and 'W'. The 'w' pauses dialing for half a second. The 'W' pauses dialing for one second. - The default for inband_on_proceeding has changed to no. Dialplan: - All channel and global variable names are evaluated in a case-sensitive manner. In previous versions of Asterisk, variables created and evaluated in the dialplan were evaluated case-insensitively, but built-in variables and variable evaluation done internally within Asterisk was done case-sensitively. - Asterisk has always had code to ignore dash '-' characters that are not part of a character set in the dialplan extensions. The code now consistently ignores these characters when matching dialplan extensions. - BRIDGE_FEATURES channel variable is now casesensitive for feature letter codes. Uppercase variants apply them to the calling party while lowercase variants apply them to the called party. 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. - In order to improve compliance with RFC 3261, SIP usernames are now properly escaped when encoding reserved characters. Prior to this change, the use of these characters in certain SIP settings affecting usernames could cause injections of these characters in their raw form into SIP headers which could in turn cause all sorts of nasty behaviors. All characters that are not alphanumeric or are not contained in the the following lists specified by RFC 3261 section 25.1 will be escaped as %XX when encoding a SIP username: * mark: "-" / "_" / "." / "!" / "~" / "*" / "'" / "(" / ")" * user-unreserved: "&" / "=" / "+" / "$" / "," / ";" / "?" / "/" 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. =========================================================== ===========================================================