2005-11-01 19:57:20 +00:00
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Information for Upgrading From Previous Asterisk Releases
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=========================================================
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2005-02-14 23:20:01 +00:00
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2006-11-19 04:22:33 +00:00
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AEL:
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* Macros are now implemented underneath with the Gosub() application.
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Heaven Help You if you wrote code depending on any aspect of this!
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Previous to 1.6, macros were implemented with the Macro() app, which
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provided a nice feature of auto-returning. The compiler will do its
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best to insert a Return() app call at the end of your macro if you did
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not include it, but really, you should make sure that all execution
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paths within your macros end in "return;".
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2007-08-15 19:21:27 +00:00
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* The conf2ael program is 'introduced' in this release; it is in a rather
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crude state, but deemed useful for making a first pass at converting
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extensions.conf code into AEL. More intelligence will come with time.
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2006-12-27 21:18:27 +00:00
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Core:
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* The 'languageprefix' option in asterisk.conf is now deprecated, and
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the default sound file layout for non-English sounds is the 'new
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style' layout introduced in Asterisk 1.4 (and used by the automatic
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sound file installer in the Makefile).
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2006-12-31 04:54:20 +00:00
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2007-07-02 21:50:15 +00:00
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* The ast_expr2 stuff has been modified to handle floating-point numbers.
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Numbers of the format D.D are now acceptable input for the expr parser,
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Where D is a string of base-10 digits. All math is now done in "long double",
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if it is available on your compiler/architecture. This was half-way between
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a bug-fix (because the MATH func returns fp by default), and an enhancement.
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In regards to changes for 9508, expr2 system choking on floating point numbers, I'm adding this update to round out (no pun intended) and make this FP-capable version of the Expr2 stuff interoperate better with previous integer-only usage, by providing Functions syntax, with 20 builtin functions for floating pt to integer conversions, and some general floating point math routines that might commonly be used also. Along with this, I made it so if a function was not a builtin, it will try and find it in the ast_custom_function list, and if found, execute it and collect the results. Thus, you can call system functions like CDR(), CHANNEL(), etc, from within $\[..\] exprs, without having to wrap them in $\{...\} (curly brace) notation. Did a valgrind on the standalone and made sure there's no mem leaks. Looks good. Updated the docs, too.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-05 18:15:22 +00:00
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Also, for those counting on, or needing, integer operations, a series of
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'functions' were also added to the expr language, to allow several styles
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of rounding/truncation, along with a set of common floating point operations,
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like sin, cos, tan, log, pow, etc. The ability to call external functions
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like CDR(), etc. was also added, without having to use the ${...} notation.
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2007-07-02 21:50:15 +00:00
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2007-07-23 19:51:41 +00:00
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* The delimiter passed to applications has been changed to the comma (','), as
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that is what people are used to using within extensions.conf. If you are
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using realtime extensions, you will need to translate your existing dialplan
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to use this separator. To use a literal comma, you need merely to escape it
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with a backslash ('\'). Another possible side effect is that you may need to
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remove the obscene level of backslashing that was necessary for the dialplan
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to work correctly in 1.4 and previous versions. This should make writing
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dialplans less painful in the future, albeit with the pain of a one-time
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conversion.
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2007-08-28 20:03:48 +00:00
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* The logger.conf option 'rotatetimestamp' has been deprecated in favor of
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'rotatestrategy'. This new option supports a 'rotate' strategy that more
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closely mimics the system logger in terms of file rotation.
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2007-11-26 19:24:23 +00:00
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* The concise versions of various CLI commands are now deprecated. We recommend
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using the manager interface (AMI) for application integration with Asterisk.
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2007-04-28 21:01:44 +00:00
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Voicemail:
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2006-12-31 04:54:20 +00:00
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* The voicemail configuration values 'maxmessage' and 'minmessage' have
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been changed to 'maxsecs' and 'minsecs' to clarify their purpose and
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to make them more distinguishable from 'maxmsgs', which sets folder
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size. The old variables will continue to work in this version, albeit
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with a deprecation warning.
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2007-04-28 21:01:44 +00:00
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* If you use any interface for modifying voicemail aside from the built in
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dialplan applications, then the option "pollmailboxes" *must* be set in
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voicemail.conf for message waiting indication (MWI) to work properly. This
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is because Voicemail notification is now event based instead of polling
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based. The channel drivers are no longer responsible for constantly manually
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checking mailboxes for changes so that they can send MWI information to users.
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Examples of situations that would require this option are web interfaces to
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voicemail or an email client in the case of using IMAP storage.
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Applications:
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2007-01-24 20:35:54 +00:00
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* ChanIsAvail() now has a 't' option, which allows the specified device
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to be queried for state without consulting the channel drivers. This
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performs mostly a 'ChanExists' sort of function.
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2007-02-03 22:06:46 +00:00
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* SetCallerPres() has been replaced with the CALLERPRES() dialplan function
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and is now deprecated.
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2007-06-11 22:21:28 +00:00
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* DISA()'s fifth argument is now an options argument. If you have previously
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used 'NOANSWER' in this argument, you'll need to convert that to the new
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option 'n'.
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2007-07-31 16:44:25 +00:00
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* Macro() is now deprecated. If you need subroutines, you should use the
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Gosub()/Return() applications. To replace MacroExclusive(), we have
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introduced dialplan functions LOCK(), TRYLOCK(), and UNLOCK(). You may use
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these functions in any location where you desire to ensure that only one
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channel is executing that path at any one time.
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2007-11-21 18:38:18 +00:00
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* Read() now sets a READSTATUS variable on exit. It does NOT automatically
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return -1 (and hangup) anymore on error. If you want to hangup on error,
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you need to do so explicitly in your dialplan.
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2007-12-14 19:27:54 +00:00
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* Privacy() no longer uses privacy.conf, so any options must be specified
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directly in the application arguments.
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2007-03-13 21:22:33 +00:00
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2007-10-29 20:13:23 +00:00
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Dialplan Functions:
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* QUEUE_MEMBER_COUNT() has been deprecated in favor of the QUEUE_MEMBER() function. For
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more information, issue a "show function QUEUE_MEMBER" from the CLI.
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2007-03-13 21:22:33 +00:00
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CDR:
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* The cdr_sqlite module has been marked as deprecated in favor of
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cdr_sqlite3_custom. It will potentially be removed from the tree
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after Asterisk 1.6 is released.
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2007-04-09 16:30:04 +00:00
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2007-11-01 22:43:46 +00:00
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* The cdr_odbc module now uses res_odbc to manage its connections. The
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username and password parameters in cdr_odbc.conf, therefore, are no
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longer used. The dsn parameter now points to an entry in res_odbc.conf.
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2007-04-09 16:30:04 +00:00
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Formats:
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* format_wav: The GAIN preprocessor definition and source code that used it
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is removed. This change was made in response to user complaints of
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2007-11-15 15:39:03 +00:00
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choppiness or the clipping of loud signal peaks. To increase the volume
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2007-04-09 16:30:04 +00:00
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of voicemail messages, use the 'volgain' option in voicemail.conf
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2007-05-04 16:37:23 +00:00
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Channel Drivers:
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2007-11-26 19:24:23 +00:00
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* SIP: a small upgrade to support the "Record" button on the SNOM360,
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2007-05-04 16:37:23 +00:00
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which sends a sip INFO message with a "Record: on" or "Record: off"
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2007-11-26 19:24:23 +00:00
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header. If Asterisk is set up (via features.conf) to accept "One Touch Monitor"
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2007-05-04 16:37:23 +00:00
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requests (by default, via '*1'), then the user-configured dialpad sequence
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is generated, and recording can be started and stopped via this button. The
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file names and formats are all controlled via the normal mechanisms. If the
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user has not configured the automon feature, the normal "415 Unsupported media type"
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is returned, and nothing is done.
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2007-11-26 19:24:23 +00:00
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* SIP: The "call-limit" option is marked as deprecated. It still works in this version of
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Asterisk, but will be removed in the following version. Please use the groupcount functions
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in the dialplan to enforce call limits. The "limitonpeer" configuration option is
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now renamed to "counteronpeer".
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2007-12-05 13:09:47 +00:00
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* SIP: The "username" option is now renamed to "defaultuser" to match "defaultip".
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These are used only before registration to call a peer with the uri
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sip:defaultuser@defaultip
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The "username" setting still work, but is deprecated and will not work in
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the next version of Asterisk.
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2007-05-04 16:37:23 +00:00
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2007-07-23 20:27:26 +00:00
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* chan_local.c: the comma delimiter inside the channel name has been changed to a
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semicolon, in order to make the Local channel driver compatible with the comma
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delimiter change in applications.
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2007-12-16 10:51:53 +00:00
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* H323: The "tos" setting has changed name to "tos_audio" and "cos" to "cos_audio"
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to be compatible with settings in sip.conf. The "tos" and "cos" configuration
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is deprecated and will stop working in the next release of Asterisk.
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2007-07-23 20:27:26 +00:00
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2008-01-05 21:35:54 +00:00
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* Console: A new console channel driver, chan_console, has been added to Asterisk.
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This new module can not be loaded at the same time as chan_alsa or chan_oss. The
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default modules.conf only loads one of them (chan_oss by default). So, unless you
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have modified your modules.conf to not use the autoload option, then you will need
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to modify modules.conf to add another "noload" line to ensure that only one of
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these three modules gets loaded.
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2007-10-15 23:20:40 +00:00
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Configuration:
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* pbx_dundi.c: tos parameter changed to use new values. Old values like lowdelay,
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lowcost and other is not acceptable now. Look into qos.tex for description of
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this parameter.
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2007-09-10 20:03:51 +00:00
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Manager:
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2007-12-06 15:56:58 +00:00
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* Manager has been upgraded to version 1.1 with a lot of changes.
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Please check doc/manager_1_1.txt for information
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2007-10-15 23:20:40 +00:00
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2007-09-10 20:03:51 +00:00
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* The IAXpeers command output has been changed to more closely resemble the
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output of the SIPpeers command.
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2007-12-05 16:46:47 +00:00
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* cdr_manager now reports at the "cdr" level, not at "call" You may need to
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change your manager.conf to add the level to existing AMI users, if they
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want to see the CDR events generated.
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