mirror of https://github.com/asterisk/asterisk
https://origsvn.digium.com/svn/asterisk/trunk ........ r157706 | kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5 lines make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@157719 65c4cc65-6c06-0410-ace0-fbb531ad65f31.6.1
Kevin P. Fleming
17 years ago
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f21824e65a
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=========================================================
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=== Information for upgrading from Asterisk 1.4 to 1.6
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===
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===
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=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
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=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
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=== UPGRADE.txt -- Upgrade info for 1.4 to 1.6
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=========================================================
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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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* 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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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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* 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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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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* 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. If you would like to avoid this conversion immediately, set
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pbx_realtime=1.4 in the [compat] section of asterisk.conf. After
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transitioning, set pbx_realtime=1.6 in the same section.
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* For the same purpose as above, you may set res_agi=1.4 in the [compat]
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section of asterisk.conf to continue to use the '|' delimiter in the EXEC
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arguments of AGI applications. After converting to use the ',' delimiter,
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change this option to res_agi=1.6.
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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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* 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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Voicemail:
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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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* 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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* 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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* ChannelRedirect() will not terminate the channel that fails to do a
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channelredirect as it has done previously. Instead CHANNELREDIRECT_STATUS
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will reflect if the attempt was successful of not.
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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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* 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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* 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. The Macro() applications
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are deprecated for performance reasons. However, since Macro() has been
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around for a long time and so many dialplans depend heavily on it, for the
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sake of backwards compatibility it will not be removed . It is also worth
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noting that using both Macro() and GoSub() at the same time is _heavily_
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discouraged.
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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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* 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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* MusicOnHold application now has duration parameter which allows specifying
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timeout in seconds.
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* WaitMusicOnHold application is now deprecated in favor of extended MusicOnHold.
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* SetMusicOnHold is now deprecated. You should use Set(CHANNEL(musicclass)=...)
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instead.
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* The arguments in ExecIf changed a bit, to be more like other applications.
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The syntax is now ExecIf(<cond>?appiftrue(args):appiffalse(args)).
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* The behavior of the Set application now depends upon a compatibility option,
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set in asterisk.conf. To use the old 1.4 behavior, which allowed Set to take
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multiple key/value pairs, set app_set=1.4 in [compat] in asterisk.conf. To
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use the new behavior, which permits variables to be set with embedded commas,
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set app_set=1.6 in [compat] in asterisk.conf. Note that you can have both
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behaviors at the same time, if you switch to using MSet if you want the old
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behavior.
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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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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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* 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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* The uniqueid field in the core Asterisk structure has been changed from a
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maximum 31 character field to a 149 character field, to account for all
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possible values the systemname prefix could be. In the past, if the
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systemname was too long, the uniqueid would have been truncated.
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* The cdr_tds module now supports all versions of FreeTDS that contain
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the db-lib frontend. It will also now log the userfield variable if
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the target database table contains a column for it.
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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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choppiness or the clipping of loud signal peaks. To increase the volume
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of voicemail messages, use the 'volgain' option in voicemail.conf
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Channel Drivers:
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* SIP: a small upgrade to support the "Record" button on the SNOM360,
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which sends a sip INFO message with a "Record: on" or "Record: off"
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header. If Asterisk is set up (via features.conf) to accept "One Touch Monitor"
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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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* 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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* 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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* 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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* 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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* 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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* DAHDI: The chan_zap module that supported PSTN interfaces using
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Zaptel has been renamed to chan_dahdi, and only supports the DAHDI
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telephony driver package for PSTN interfaces. See the
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Zaptel-to-DAHDI.txt file for more details on this transition.
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* DAHDI: The "msdstrip" option has been deprecated, as it provides no value over
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the method of stripping digits in the dialplan using variable substring syntax.
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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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* queues.conf: the queue-lessthan sound file option is no longer available, and the
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queue-round-seconds option no longer takes '1' as a valid parameter.
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Manager:
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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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* 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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* 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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* The Originate command now requires the Originate write permission. For
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Originate with the Application parameter, you need the additional System
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privilege if you want to do anything that calls out to a subshell.
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iLBC Codec:
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* Previously, the Asterisk source code distribution included the iLBC
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encoder/decoder source code, from Global IP Solutions
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(http://www.gipscorp.com). This code is not licensed for
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distribution, and thus has been removed from the Asterisk source
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code distribution. If you wish to use codec_ilbc to support iLBC
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channels in Asterisk, you can run the contrib/scripts/get_ilbc_source.sh
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script to download the source and put it in the proper place in
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the Asterisk build tree. Once that is done you can follow your normal
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steps of building Asterisk. You will need to run 'menuselect' and enable
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the iLBC codec in the 'Codec Translators' category.
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