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diff --git a/UPGRADE-1.6.txt b/UPGRADE-1.6.txt new file mode 100644 index 000000000..8fb0a0f71 --- /dev/null +++ b/UPGRADE-1.6.txt @@ -0,0 +1,250 @@ +========================================================= +=== Information for upgrading from Asterisk 1.4 to 1.6 +=== +=== +=== 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.txt -- Upgrade info for 1.4 to 1.6 +========================================================= + +AEL: + +* Macros are now implemented underneath with the Gosub() application. + Heaven Help You if you wrote code depending on any aspect of this! + Previous to 1.6, macros were implemented with the Macro() app, which + provided a nice feature of auto-returning. The compiler will do its + best to insert a Return() app call at the end of your macro if you did + not include it, but really, you should make sure that all execution + paths within your macros end in "return;". + +* The conf2ael program is 'introduced' in this release; it is in a rather + crude state, but deemed useful for making a first pass at converting + extensions.conf code into AEL. More intelligence will come with time. + +Core: + +* The 'languageprefix' option in asterisk.conf is now deprecated, and + the default sound file layout for non-English sounds is the 'new + style' layout introduced in Asterisk 1.4 (and used by the automatic + sound file installer in the Makefile). + +* The ast_expr2 stuff has been modified to handle floating-point numbers. + Numbers of the format D.D are now acceptable input for the expr parser, + Where D is a string of base-10 digits. All math is now done in "long double", + if it is available on your compiler/architecture. This was half-way between + a bug-fix (because the MATH func returns fp by default), and an enhancement. + Also, for those counting on, or needing, integer operations, a series of + 'functions' were also added to the expr language, to allow several styles + of rounding/truncation, along with a set of common floating point operations, + like sin, cos, tan, log, pow, etc. The ability to call external functions + like CDR(), etc. was also added, without having to use the ${...} notation. + +* The delimiter passed to applications has been changed to the comma (','), as + that is what people are used to using within extensions.conf. If you are + using realtime extensions, you will need to translate your existing dialplan + to use this separator. To use a literal comma, you need merely to escape it + with a backslash ('\'). Another possible side effect is that you may need to + remove the obscene level of backslashing that was necessary for the dialplan + to work correctly in 1.4 and previous versions. This should make writing + dialplans less painful in the future, albeit with the pain of a one-time + conversion. If you would like to avoid this conversion immediately, set + pbx_realtime=1.4 in the [compat] section of asterisk.conf. After + transitioning, set pbx_realtime=1.6 in the same section. + +* For the same purpose as above, you may set res_agi=1.4 in the [compat] + section of asterisk.conf to continue to use the '|' delimiter in the EXEC + arguments of AGI applications. After converting to use the ',' delimiter, + change this option to res_agi=1.6. + +* The logger.conf option 'rotatetimestamp' has been deprecated in favor of + 'rotatestrategy'. This new option supports a 'rotate' strategy that more + closely mimics the system logger in terms of file rotation. + +* The concise versions of various CLI commands are now deprecated. We recommend + using the manager interface (AMI) for application integration with Asterisk. + +Voicemail: + +* The voicemail configuration values 'maxmessage' and 'minmessage' have + been changed to 'maxsecs' and 'minsecs' to clarify their purpose and + to make them more distinguishable from 'maxmsgs', which sets folder + size. The old variables will continue to work in this version, albeit + with a deprecation warning. + +* If you use any interface for modifying voicemail aside from the built in + dialplan applications, then the option "pollmailboxes" *must* be set in + voicemail.conf for message waiting indication (MWI) to work properly. This + is because Voicemail notification is now event based instead of polling + based. The channel drivers are no longer responsible for constantly manually + checking mailboxes for changes so that they can send MWI information to users. + Examples of situations that would require this option are web interfaces to + voicemail or an email client in the case of using IMAP storage. + +Applications: + + +* ChanIsAvail() now has a 't' option, which allows the specified device + to be queried for state without consulting the channel drivers. This + performs mostly a 'ChanExists' sort of function. + +* ChannelRedirect() will not terminate the channel that fails to do a + channelredirect as it has done previously. Instead CHANNELREDIRECT_STATUS + will reflect if the attempt was successful of not. + +* SetCallerPres() has been replaced with the CALLERPRES() dialplan function + and is now deprecated. + +* DISA()'s fifth argument is now an options argument. If you have previously + used 'NOANSWER' in this argument, you'll need to convert that to the new + option 'n'. + +* Macro() is now deprecated. If you need subroutines, you should use the + Gosub()/Return() applications. To replace MacroExclusive(), we have + introduced dialplan functions LOCK(), TRYLOCK(), and UNLOCK(). You may use + these functions in any location where you desire to ensure that only one + channel is executing that path at any one time. The Macro() applications + are deprecated for performance reasons. However, since Macro() has been + around for a long time and so many dialplans depend heavily on it, for the + sake of backwards compatibility it will not be removed . It is also worth + noting that using both Macro() and GoSub() at the same time is _heavily_ + discouraged. + +* Read() now sets a READSTATUS variable on exit. It does NOT automatically + return -1 (and hangup) anymore on error. If you want to hangup on error, + you need to do so explicitly in your dialplan. + +* Privacy() no longer uses privacy.conf, so any options must be specified + directly in the application arguments. + +* MusicOnHold application now has duration parameter which allows specifying + timeout in seconds. + +* WaitMusicOnHold application is now deprecated in favor of extended MusicOnHold. + +* SetMusicOnHold is now deprecated. You should use Set(CHANNEL(musicclass)=...) + instead. + +* The arguments in ExecIf changed a bit, to be more like other applications. + The syntax is now ExecIf(<cond>?appiftrue(args):appiffalse(args)). + +* The behavior of the Set application now depends upon a compatibility option, + set in asterisk.conf. To use the old 1.4 behavior, which allowed Set to take + multiple key/value pairs, set app_set=1.4 in [compat] in asterisk.conf. To + use the new behavior, which permits variables to be set with embedded commas, + set app_set=1.6 in [compat] in asterisk.conf. Note that you can have both + behaviors at the same time, if you switch to using MSet if you want the old + behavior. + +Dialplan Functions: + +* QUEUE_MEMBER_COUNT() has been deprecated in favor of the QUEUE_MEMBER() function. For + more information, issue a "show function QUEUE_MEMBER" from the CLI. + +CDR: + +* The cdr_sqlite module has been marked as deprecated in favor of + cdr_sqlite3_custom. It will potentially be removed from the tree + after Asterisk 1.6 is released. + +* The cdr_odbc module now uses res_odbc to manage its connections. The + username and password parameters in cdr_odbc.conf, therefore, are no + longer used. The dsn parameter now points to an entry in res_odbc.conf. + +* The uniqueid field in the core Asterisk structure has been changed from a + maximum 31 character field to a 149 character field, to account for all + possible values the systemname prefix could be. In the past, if the + systemname was too long, the uniqueid would have been truncated. + +* The cdr_tds module now supports all versions of FreeTDS that contain + the db-lib frontend. It will also now log the userfield variable if + the target database table contains a column for it. + +Formats: + +* format_wav: The GAIN preprocessor definition and source code that used it + is removed. This change was made in response to user complaints of + choppiness or the clipping of loud signal peaks. To increase the volume + of voicemail messages, use the 'volgain' option in voicemail.conf + +Channel Drivers: + +* SIP: a small upgrade to support the "Record" button on the SNOM360, + which sends a sip INFO message with a "Record: on" or "Record: off" + header. If Asterisk is set up (via features.conf) to accept "One Touch Monitor" + requests (by default, via '*1'), then the user-configured dialpad sequence + is generated, and recording can be started and stopped via this button. The + file names and formats are all controlled via the normal mechanisms. If the + user has not configured the automon feature, the normal "415 Unsupported media type" + is returned, and nothing is done. + +* SIP: The "call-limit" option is marked as deprecated. It still works in this version of + Asterisk, but will be removed in the following version. Please use the groupcount functions + in the dialplan to enforce call limits. The "limitonpeer" configuration option is + now renamed to "counteronpeer". + +* SIP: The "username" option is now renamed to "defaultuser" to match "defaultip". + These are used only before registration to call a peer with the uri + sip:defaultuser@defaultip + The "username" setting still work, but is deprecated and will not work in + the next version of Asterisk. + +* chan_local.c: the comma delimiter inside the channel name has been changed to a + semicolon, in order to make the Local channel driver compatible with the comma + delimiter change in applications. + +* H323: The "tos" setting has changed name to "tos_audio" and "cos" to "cos_audio" + to be compatible with settings in sip.conf. The "tos" and "cos" configuration + is deprecated and will stop working in the next release of Asterisk. + +* Console: A new console channel driver, chan_console, has been added to Asterisk. + This new module can not be loaded at the same time as chan_alsa or chan_oss. The + default modules.conf only loads one of them (chan_oss by default). So, unless you + have modified your modules.conf to not use the autoload option, then you will need + to modify modules.conf to add another "noload" line to ensure that only one of + these three modules gets loaded. + +* DAHDI: The chan_zap module that supported PSTN interfaces using + Zaptel has been renamed to chan_dahdi, and only supports the DAHDI + telephony driver package for PSTN interfaces. See the + Zaptel-to-DAHDI.txt file for more details on this transition. + +* DAHDI: The "msdstrip" option has been deprecated, as it provides no value over + the method of stripping digits in the dialplan using variable substring syntax. + +Configuration: + +* pbx_dundi.c: tos parameter changed to use new values. Old values like lowdelay, + lowcost and other is not acceptable now. Look into qos.tex for description of + this parameter. + +* queues.conf: the queue-lessthan sound file option is no longer available, and the + queue-round-seconds option no longer takes '1' as a valid parameter. + +Manager: + +* Manager has been upgraded to version 1.1 with a lot of changes. + Please check doc/manager_1_1.txt for information + +* The IAXpeers command output has been changed to more closely resemble the + output of the SIPpeers command. + +* cdr_manager now reports at the "cdr" level, not at "call" You may need to + change your manager.conf to add the level to existing AMI users, if they + want to see the CDR events generated. + +* The Originate command now requires the Originate write permission. For + Originate with the Application parameter, you need the additional System + privilege if you want to do anything that calls out to a subshell. + +iLBC Codec: + +* Previously, the Asterisk source code distribution included the iLBC + encoder/decoder source code, from Global IP Solutions + (http://www.gipscorp.com). This code is not licensed for + distribution, and thus has been removed from the Asterisk source + code distribution. If you wish to use codec_ilbc to support iLBC + channels in Asterisk, you can run the contrib/scripts/get_ilbc_source.sh + script to download the source and put it in the proper place in + the Asterisk build tree. Once that is done you can follow your normal + steps of building Asterisk. You will need to run 'menuselect' and enable + the iLBC codec in the 'Codec Translators' category. |