196 lines
9.5 KiB
Plaintext
196 lines
9.5 KiB
Plaintext
Information for Upgrading From Previous Asterisk Releases
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=========================================================
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Build Process (configure script):
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Asterisk now uses an autoconf-generated configuration script to learn how it
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should build itself for your system. As it is a standard script, running:
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$ ./configure --help
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will show you all the options available. This script can be used to tell the
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build process what libraries you have on your system (if it cannot find them
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automatically), which libraries you wish to have ignored even though they may
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be present, etc.
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You must run the configure script before Asterisk will build, although it will
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attempt to automatically run it for you with no options specified; for most users,
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that will result in a similar build to what they would have had before the
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configure script was added to the build process (except for having to run 'make'
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again after the configure script is run). Note that the configure script does NOT
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need to be re-run just to rebuild Asterisk; you only need to re-run it when your
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system configuration changes or you wish to build Asterisk with different options.
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Build Process (module selection):
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The Asterisk source tree now includes a basic module selection and build option
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selection tool called 'menuselect'. Run 'make menuselect' to make your choices.
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In this tool, you can disable building of modules that you don't care about,
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turn on/off global options for the build and see which modules will not (and cannot)
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be built because your system does not have the required external dependencies
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installed.
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(TODO: document where 'global' and 'per-user' menuselect input files should go
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and what they need to contain)
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PBX Core:
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* The (very old and undocumented) ability to use BYEXTENSION for dialing
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instead of ${EXTEN} has been removed.
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Command Line Interface:
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* 'show channels concise', designed to be used by applications that will parse
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its output, previously used ':' characters to separate fields. However, some
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of those fields can easily contain that character, making the output not
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parseable. The delimiter has been changed to '!'.
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Applications:
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* In previous Asterisk releases, many applications would jump to priority n+101
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to indicate some kind of status or error condition. This functionality was
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marked deprecated in Asterisk 1.2. An option to disable it was provided with
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the default value set to 'on'. The default value for the global priority
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jumping option is now 'off'.
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* The applications Cut, Sort, DBGet, DBPut, SetCIDNum, SetCIDName, SetRDNIS,
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AbsoluteTimeout, DigitTimeout, ResponseTimeout, SetLanguage, GetGroupCount,
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and GetGroupMatchCount were all deprecated in version 1.2, and therefore have
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been removed in this version. You should use the equivalent dialplan
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function in places where you have previously used one of these applications.
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* The application SetVar has been renamed to Set. The syntax SetVar was marked
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deprecated in version 1.2 and is no longer recognized in this version.
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* app_read has been updated to use the newer options codes, using "skip" or
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"noanswer" will not work. Use s or n. Also there is a new feature i, for
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using indication tones, so typing in skip would give you unexpected results.
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* OSPAuth is added to authenticate OSP tokens in in_bound call setup messages.
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* The CONNECT event in the queue_log from app_queue now has a second field
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in addition to the holdtime field. It contains the unique ID of the
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queue member channel that is taking the call. This is useful when trying
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to link recording filenames back to a particular call from the queue.
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* The old/current behavior of app_queue has a serial type behavior
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in that the queue will make all waiting callers wait in the queue
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even if there is more than one available member ready to take
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calls until the head caller is connected with the member they
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were trying to get to. The next waiting caller in line then
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becomes the head caller, and they are then connected with the
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next available member and all available members and waiting callers
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waits while this happens. This cycle continues until there are
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no more available members or waiting callers, whichever comes first.
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The new behavior, enabled by setting autofill=yes in queues.conf
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either at the [general] level to default for all queues or
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to set on a per-queue level, makes sure that when the waiting
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callers are connecting with available members in a parallel fashion
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until there are no more available members or no more waiting callers,
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whichever comes first. This is probably more along the lines of how
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one would expect a queue should work and in most cases, you will want
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to enable this new behavior. If you do not specify or comment out this
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option, it will default to "no" to keep backward compatability with the old
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behavior.
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* The app_queue application now has the ability to use MixMonitor to
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record conversations queue members are having with queue callers. Please
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see configs/queues.conf.sample for more information on this option.
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* ast_play_and_record would attempt to cancel the recording if a DTMF
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'0' was received. This behavior was not documented in most of the
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applications that used ast_play_and_record and the return codes from
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ast_play_and_record weren't checked for properly.
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ast_play_and_record has been changed so that '0' no longer cancels a
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recording. If you want to allow DTMF digits to cancel an
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in-progress recording use ast_play_and_record_full which allows you
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to specify which DTMF digits can be used to accept a recording and
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which digits can be used to cancel a recording.
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Manager:
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* After executing the 'status' manager action, the "Status" manager events
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included the header "CallerID:" which was actually only the CallerID number,
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and not the full CallerID string. This header has been renamed to
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"CallerIDNum". For compatibility purposes, the CallerID parameter will remain
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until after the release of 1.4, when it will be removed. Please use the time
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during the 1.4 release to make this transition.
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* The AgentConnect event now has an additional field called "BridgedChannel"
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which contains the unique ID of the queue member channel that is taking the
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call. This is useful when trying to link recording filenames back to
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a particular call from the queue.
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Variables:
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* The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
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${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE},
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and ${LANGUAGE} have all been deprecated in favor of their related dialplan
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functions. You are encouraged to move towards the associated dialplan
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function, as these variables will be removed in a future release.
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* The CDR-CSV variables uniqueid, userfield, and basing time on GMT are now
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adjustable from cdr.conf, instead of recompiling.
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* OSP applications exports several new variables, ${OSPINHANDLE},
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${OSPOUTHANDLE}, ${OSPINTOKEN}, ${OSPOUTTOKEN}, ${OSPCALLING},
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${OSPINTIMELIMIT}, and ${OSPOUTTIMELIMIT}
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Functions:
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* The function ${CHECK_MD5()} has been deprecated in favor of using an
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expression: $[${MD5(<string>)} = ${saved_md5}].
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* The 'builtin' functions that used to be combined in pbx_functions.so are
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now built as separate modules. If you are not using 'autoload=yes' in your
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modules.conf file then you will need to explicitly load the modules that
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contain the functions you want to use.
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* The ENUMLOOKUP() function with the 'c' option (for counting the number of records),
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but the lookup fails to match any records, the returned value will now be "0" instead of blank.
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* The REALTIME() function is now available in version 1.4 and app_realtime has
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been deprecated in favor of the new function. app_realtime will be removed
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completely with the version 1.6 release so please take the time between
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releases to make any necessary changes
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The IAX2 channel:
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* The "mailboxdetail" option has been deprecated. Previously, if this option
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was not enabled, the 2 byte MSGCOUNT information element would be set to all
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1's to indicate there there is some number of messages waiting. With this
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option enabled, the number of new messages were placed in one byte and the
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number of old messages are placed in the other. This is now the default
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(and the only) behavior.
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The SIP channel:
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* The "incominglimit" setting is replaced by the "call-limit" setting in sip.conf.
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* OSP support code is removed from SIP channel to OSP applications. ospauth
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option in sip.conf is removed to osp.conf as authpolicy. allowguest option
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in sip.conf cannot be set as osp anymore.
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The Zap channel:
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* Support for MFC/R2 has been removed, as it has not been functional for some time
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and it has no maintainer.
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Installation:
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* On BSD systems, the installation directories have changed to more "FreeBSDish" directories. On startup, Asterisk will look for the main configuration in /usr/local/etc/asterisk/asterisk.conf
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If you have an old installation, you might want to remove the binaries and move the configuration files to the new locations. The following directories are now default:
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ASTLIBDIR /usr/local/lib/asterisk
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ASTVARLIBDIR /usr/local/share/asterisk
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ASTETCDIR /usr/local/etc/asterisk
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ASTBINDIR /usr/local/bin/asterisk
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ASTSBINDIR /usr/local/sbin/asterisk
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Sounds:
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* The phonetic sounds directory has been removed from the asterisk-sounds package
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because they are now included directly in Asterisk. However, it is important to
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note that the phonetic sounds that existed in asterisk-sounds used a different
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naming convention than the sounds in Asterisk. For example, instead of alpha.gsm
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and bravo.gsm, Asterisk has a_p.gsm and b_p.gsm.
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