path: root/UPGRADE-1.6.txt
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authorkpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b>2008-11-19 13:19:49 +0000
committerkpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b>2008-11-19 13:19:49 +0000
commit84de3e7ec27cbda09e16e0c2a49e330862ac271f (patch)
tree01ab767bcc6afcdf0de8f3c318a97b5122cb720e /UPGRADE-1.6.txt
parent296683f301336b411ceef41a96ebe5a9684fc6e0 (diff)
Merged revisions 157706 via svnmerge from
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@157738 f38db490-d61c-443f-a65b-d21fe96a405b
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+=== 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
+* 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.
+* 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.
+* 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.
+* 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.
+* 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.
+* 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.
+* 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 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.