path: root/UPGRADE-1.4.txt
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authorrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2008-02-08 16:49:19 +0000
committerrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2008-02-08 16:49:19 +0000
commitdc7deea755bcd816c592bcb8b2ece9833ccd3286 (patch)
tree3e5887abbf82d22867b18d38b262bd34bf82dcc6 /UPGRADE-1.4.txt
parentea77bf657eb475eabf1bebe756714f1018db9caa (diff)
At the request of ManxPower, include the UPGRADE.txt from 1.2 and 1.4, as well.
This way, if people need to go back and review what was deprecated in previous major releases, it is readily available to them. Thanks for the suggestion! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103044 f38db490-d61c-443f-a65b-d21fe96a405b
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+=== Information for upgrading from Asterisk 1.2 to 1.4
+=== 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
+Build Process (configure script):
+Asterisk now uses an autoconf-generated configuration script to learn how it
+should build itself for your system. As it is a standard script, running:
+$ ./configure --help
+will show you all the options available. This script can be used to tell the
+build process what libraries you have on your system (if it cannot find them
+automatically), which libraries you wish to have ignored even though they may
+be present, etc.
+You must run the configure script before Asterisk will build, although it will
+attempt to automatically run it for you with no options specified; for most
+users, that will result in a similar build to what they would have had before
+the configure script was added to the build process (except for having to run
+'make' again after the configure script is run). Note that the configure script
+does NOT need to be re-run just to rebuild Asterisk; you only need to re-run it
+when your system configuration changes or you wish to build Asterisk with
+different options.
+Build Process (module selection):
+The Asterisk source tree now includes a basic module selection and build option
+selection tool called 'menuselect'. Run 'make menuselect' to make your choices.
+In this tool, you can disable building of modules that you don't care about,
+turn on/off global options for the build and see which modules will not
+(and cannot) be built because your system does not have the required external
+dependencies installed.
+The resulting file from menuselect is called 'menuselect.makeopts'. Note that
+the resulting menuselect.makeopts file generally contains which modules *not*
+to build. The modules listed in this file indicate which modules have unmet
+dependencies, a present conflict, or have been disabled by the user in the
+menuselect interface. Compiler Flags can also be set in the menuselect
+interface. In this case, the resulting file contains which CFLAGS are in use,
+not which ones are not in use.
+If you would like to save your choices and have them applied against all
+builds, the file can be copied to '~/.asterisk.makeopts' or
+Build Process (Makefile targets):
+The 'valgrind' and 'dont-optimize' targets have been removed; their functionality
+is available by enabling the DONT_OPTIMIZE setting in the 'Compiler Flags' menu
+in the menuselect tool.
+It is now possible to run most make targets against a single subdirectory; from
+the top level directory, for example, 'make channels' will run 'make all' in the
+'channels' subdirectory. This also is true for 'clean', 'distclean' and 'depend'.
+Sound (prompt) and Music On Hold files:
+Beginning with Asterisk 1.4, the sound files and music on hold files supplied for
+use with Asterisk have been replaced with new versions produced from high quality
+master recordings, and are available in three languages (English, French and
+Spanish) and in five formats (WAV (uncompressed), mu-Law, a-Law, GSM and G.729).
+In addition, the music on hold files provided by FreePlay Music are now available
+in the same five formats, but no longer available in MP3 format.
+The Asterisk 1.4 tarball packages will only include English prompts in GSM format,
+(as were supplied with previous releases) and the FreePlay MOH files in WAV format.
+All of the other variations can be installed by running 'make menuselect' and
+selecting the packages you wish to install; when you run 'make install', those
+packages will be downloaded and installed along with the standard files included
+in the tarball.
+If for some reason you expect to not have Internet access at the time you will be
+running 'make install', you can make your package selections using menuselect and
+then run 'make sounds' to download (only) the sound packages; this will leave the
+sound packages in the 'sounds' subdirectory to be used later during installation.
+WARNING: Asterisk 1.4 supports a new layout for sound files in multiple languages;
+instead of the alternate-language files being stored in subdirectories underneath
+the existing files (for French, that would be digits/fr, letters/fr, phonetic/fr,
+etc.) the new layout creates one directory under /var/lib/asterisk/sounds for the
+language itself, then places all the sound files for that language under that
+directory and its subdirectories. This is the layout that will be created if you
+select non-English languages to be installed via menuselect, HOWEVER Asterisk does
+not default to this layout and will not find the files in the places it expects them
+to be. If you wish to use this layout, make sure you put 'languageprefix=yes' in your
+/etc/asterisk/asterisk.conf file, so that Asterisk will know how the files were
+PBX Core:
+* The (very old and undocumented) ability to use BYEXTENSION for dialing
+ instead of ${EXTEN} has been removed.
+* Builtin (res_features) transfer functionality attempts to use the context
+ defined in TRANSFER_CONTEXT variable of the transferer channel first. If
+ not set, it uses the transferee variable. If not set in any channel, it will
+ attempt to use the last non macro context. If not possible, it will default
+ to the current context.
+* The autofallthrough setting introduced in Asterisk 1.2 now defaults to 'yes';
+ if your dialplan relies on the ability to 'run off the end' of an extension
+ and wait for a new extension without using WaitExten() to accomplish that,
+ you will need set autofallthrough to 'no' in your extensions.conf file.
+Command Line Interface:
+* 'show channels concise', designed to be used by applications that will parse
+ its output, previously used ':' characters to separate fields. However, some
+ of those fields can easily contain that character, making the output not
+ parseable. The delimiter has been changed to '!'.
+* In previous Asterisk releases, many applications would jump to priority n+101
+ to indicate some kind of status or error condition. This functionality was
+ marked deprecated in Asterisk 1.2. An option to disable it was provided with
+ the default value set to 'on'. The default value for the global priority
+ jumping option is now 'off'.
+* The applications Cut, Sort, DBGet, DBPut, SetCIDNum, SetCIDName, SetRDNIS,
+ AbsoluteTimeout, DigitTimeout, ResponseTimeout, SetLanguage, GetGroupCount,
+ and GetGroupMatchCount were all deprecated in version 1.2, and therefore have
+ been removed in this version. You should use the equivalent dialplan
+ function in places where you have previously used one of these applications.
+* The application SetGlobalVar has been deprecated. You should replace uses
+ of this application with the following combination of Set and GLOBAL():
+ Set(GLOBAL(name)=value). You may also access global variables exclusively by
+ using the GLOBAL() dialplan function, instead of relying on variable
+ interpolation falling back to globals when no channel variable is set.
+* The application SetVar has been renamed to Set. The syntax SetVar was marked
+ deprecated in version 1.2 and is no longer recognized in this version. The
+ use of Set with multiple argument pairs has also been deprecated. Please
+ separate each name/value pair into its own dialplan line.
+* app_read has been updated to use the newer options codes, using "skip" or
+ "noanswer" will not work. Use s or n. Also there is a new feature i, for
+ using indication tones, so typing in skip would give you unexpected results.
+* OSPAuth is added to authenticate OSP tokens in in_bound call setup messages.
+* The CONNECT event in the queue_log from app_queue now has a second field
+ in addition to the holdtime field. It contains the unique ID of the
+ queue member channel that is taking the call. This is useful when trying
+ to link recording filenames back to a particular call from the queue.
+* The old/current behavior of app_queue has a serial type behavior
+ in that the queue will make all waiting callers wait in the queue
+ even if there is more than one available member ready to take
+ calls until the head caller is connected with the member they
+ were trying to get to. The next waiting caller in line then
+ becomes the head caller, and they are then connected with the
+ next available member and all available members and waiting callers
+ waits while this happens. This cycle continues until there are
+ no more available members or waiting callers, whichever comes first.
+ The new behavior, enabled by setting autofill=yes in queues.conf
+ either at the [general] level to default for all queues or
+ to set on a per-queue level, makes sure that when the waiting
+ callers are connecting with available members in a parallel fashion
+ until there are no more available members or no more waiting callers,
+ whichever comes first. This is probably more along the lines of how
+ one would expect a queue should work and in most cases, you will want
+ to enable this new behavior. If you do not specify or comment out this
+ option, it will default to "no" to keep backward compatability with the old
+ behavior.
+* Queues depend on the channel driver reporting the proper state
+ for each member of the queue. To get proper signalling on
+ queue members that use the SIP channel driver, you need to
+ enable a call limit (could be set to a high value so it
+ is not put into action) and also make sure that both inbound
+ and outbound calls are accounted for.
+ Example:
+ [general]
+ limitonpeer = yes
+ [peername]
+ type=friend
+ call-limit=10
+* The app_queue application now has the ability to use MixMonitor to
+ record conversations queue members are having with queue callers. Please
+ see configs/queues.conf.sample for more information on this option.
+* The app_queue application strategy called 'roundrobin' has been deprecated
+ for this release. Users are encouraged to use 'rrmemory' instead, since it
+ provides more 'true' round-robin call delivery. For the Asterisk 1.6 release,
+ 'rrmemory' will be renamed 'roundrobin'.
+* The app_queue application option called 'monitor-join' has been deprecated
+ for this release. Users are encouraged to use 'monitor-type=mixmonitor' instead,
+ since it provides the same functionality but is not dependent on soxmix or some
+ other external program in order to mix the audio.
+* app_meetme: The 'm' option (monitor) is renamed to 'l' (listen only), and
+ the 'm' option now provides the functionality of "initially muted".
+ In practice, most existing dialplans using the 'm' flag should not notice
+ any difference, unless the keypad menu is enabled, allowing the user
+ to unmute themsleves.
+* ast_play_and_record would attempt to cancel the recording if a DTMF
+ '0' was received. This behavior was not documented in most of the
+ applications that used ast_play_and_record and the return codes from
+ ast_play_and_record weren't checked for properly.
+ ast_play_and_record has been changed so that '0' no longer cancels a
+ recording. If you want to allow DTMF digits to cancel an
+ in-progress recording use ast_play_and_record_full which allows you
+ to specify which DTMF digits can be used to accept a recording and
+ which digits can be used to cancel a recording.
+* ast_app_messagecount has been renamed to ast_app_inboxcount. There is now a
+ new ast_app_messagecount function which takes a single context/mailbox/folder
+ mailbox specification and returns the message count for that folder only.
+ This addresses the deficiency of not being able to count the number of
+ messages in folders other than INBOX and Old.
+* The exit behavior of the AGI applications has changed. Previously, when
+ a connection to an AGI server failed, the application would cause the channel
+ to immediately stop dialplan execution and hangup. Now, the only time that
+ the AGI applications will cause the channel to stop dialplan execution is
+ when the channel itself requests hangup. The AGI applications now set an
+ AGISTATUS variable which will allow you to find out whether running the AGI
+ was successful or not.
+ Previously, there was no way to handle the case where Asterisk was unable to
+ locally execute an AGI script for some reason. In this case, dialplan
+ execution will continue as it did before, but the AGISTATUS variable will be
+ set to "FAILURE".
+ A locally executed AGI script can now exit with a non-zero exit code and this
+ failure will be detected by Asterisk. If an AGI script exits with a non-zero
+ exit code, the AGISTATUS variable will be set to "FAILURE" as opposed to
+* app_voicemail: The ODBC_STORAGE capability now requires the extended table format
+ previously used only by EXTENDED_ODBC_STORAGE. This means that you will need to update
+ your table format using the schema provided in doc/odbcstorage.txt
+* app_waitforsilence: Fixes have been made to this application which changes the
+ default behavior with how quickly it returns. You can maintain "old-style" behavior
+ with the addition/use of a third "timeout" parameter.
+ Please consult the application documentation and make changes to your dialplan
+ if appropriate.
+* After executing the 'status' manager action, the "Status" manager events
+ included the header "CallerID:" which was actually only the CallerID number,
+ and not the full CallerID string. This header has been renamed to
+ "CallerIDNum". For compatibility purposes, the CallerID parameter will remain
+ until after the release of 1.4, when it will be removed. Please use the time
+ during the 1.4 release to make this transition.
+* The AgentConnect event now has an additional field called "BridgedChannel"
+ which contains the unique ID of the queue member channel that is taking the
+ call. This is useful when trying to link recording filenames back to
+ a particular call from the queue.
+* app_userevent has been modified to always send Event: UserEvent with the
+ additional header UserEvent: <userspec>. Also, the Channel and UniqueID
+ headers are not automatically sent, unless you specify them as separate
+ arguments. Please see the application help for the new syntax.
+* app_meetme: Mute and Unmute events are now reported via the Manager API.
+ Native Manager API commands MeetMeMute and MeetMeUnmute are provided, which
+ are easier to use than "Action Command:". The MeetMeStopTalking event has
+ also been deprecated in favor of the already existing MeetmeTalking event
+ with a "Status" of "on" or "off" added.
+* OriginateFailure and OriginateSuccess events were replaced by event
+ OriginateResponse with a header named "Response" to indicate success or
+ failure
+* The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
+ and ${LANGUAGE} have all been deprecated in favor of their related dialplan
+ functions. You are encouraged to move towards the associated dialplan
+ function, as these variables will be removed in a future release.
+* The CDR-CSV variables uniqueid, userfield, and basing time on GMT are now
+ adjustable from cdr.conf, instead of recompiling.
+* OSP applications exports several new variables, ${OSPINHANDLE},
+* Builtin transfer functionality sets the variable ${TRANSFERERNAME} in the new
+ created channel. This variables holds the channel name of the transferer.
+* The dial plan variable PRI_CAUSE will be removed from future versions
+ of Asterisk.
+ It is replaced by adding a cause value to the hangup() application.
+* The function ${CHECK_MD5()} has been deprecated in favor of using an
+ expression: $[${MD5(<string>)} = ${saved_md5}].
+* The 'builtin' functions that used to be combined in pbx_functions.so are
+ now built as separate modules. If you are not using 'autoload=yes' in your
+ modules.conf file then you will need to explicitly load the modules that
+ contain the functions you want to use.
+* The ENUMLOOKUP() function with the 'c' option (for counting the number of
+ records), but the lookup fails to match any records, the returned value will
+ now be "0" instead of blank.
+* The REALTIME() function is now available in version 1.4 and app_realtime has
+ been deprecated in favor of the new function. app_realtime will be removed
+ completely with the version 1.6 release so please take the time between
+ releases to make any necessary changes
+* The QUEUEAGENTCOUNT() function has been deprecated in favor of
+The IAX2 channel:
+* It is possible that previous configurations depended on the order in which
+ peers and users were specified in iax.conf for forcing the order in which
+ chan_iax2 matched against them. This behavior is going away and is considered
+ deprecated in this version. Avoid having ambiguous peer and user entries and
+ to make things easy on yourself, always set the "username" option for users
+ so that the remote end can match on that exactly instead of trying to infer
+ which user you want based on host.
+ If you would like to go ahead and use the new behavior which doesn't use the
+ order in the config file to influence matching order, then change the
+ MAX_PEER_BUCKETS define in chan_iax2.c to a value greater than one. An
+ example is provided there. By changing this, you will get *much* better
+ performance on systems that do a lot of peer and user lookups as they will be
+ stored in memory in a much more efficient manner.
+* The "mailboxdetail" option has been deprecated. Previously, if this option
+ was not enabled, the 2 byte MSGCOUNT information element would be set to all
+ 1's to indicate there there is some number of messages waiting. With this
+ option enabled, the number of new messages were placed in one byte and the
+ number of old messages are placed in the other. This is now the default
+ (and the only) behavior.
+The SIP channel:
+* The "incominglimit" setting is replaced by the "call-limit" setting in
+ sip.conf.
+* OSP support code is removed from SIP channel to OSP applications. ospauth
+ option in sip.conf is removed to osp.conf as authpolicy. allowguest option
+ in sip.conf cannot be set as osp anymore.
+* The Asterisk RTP stack has been changed in regards to RFC2833 reception
+ and transmission. Packets will now be sent with proper duration instead of all
+ at once. If you are receiving calls from a pre-1.4 Asterisk installation you
+ will want to turn on the rfc2833compensate option. Without this option your
+ DTMF reception may act poorly.
+* The $SIPUSERAGENT dialplan variable is deprecated and will be removed
+ in coming versions of Asterisk. Please use the dialplan function
+ SIPCHANINFO(useragent) instead.
+* The ALERT_INFO dialplan variable is deprecated and will be removed
+ in coming versions of Asterisk. Please use the dialplan application
+ sipaddheader() to add the "Alert-Info" header to the outbound invite.
+* The "canreinvite" option has changed. canreinvite=yes used to disable
+ re-invites if you had NAT=yes. In 1.4, you need to set canreinvite=nonat
+ to disable re-invites when NAT=yes. This is propably what you want.
+ The settings are now: "yes", "no", "nonat", "update". Please consult
+ sip.conf.sample for detailed information.
+The Zap channel:
+* Support for MFC/R2 has been removed, as it has not been functional for some
+ time and it has no maintainer.
+The Agent channel:
+* Callback mode (AgentCallbackLogin) is now deprecated, since the entire function
+ it provided can be done using dialplan logic, without requiring additional
+ channel and module locks (which frequently caused deadlocks). An example of
+ how to do this using AEL dialplan is in doc/queues-with-callback-members.txt.
+The G726-32 codec:
+* It has been determined that previous versions of Asterisk used the wrong codeword
+ packing order for G726-32 data. This version supports both available packing orders,
+ and can transcode between them. It also now selects the proper order when
+ negotiating with a SIP peer based on the codec name supplied in the SDP. However,
+ there are existing devices that improperly request one order and then use another;
+ Sipura and Grandstream ATAs are known to do this, and there may be others. To
+ be able to continue to use these devices with this version of Asterisk and the
+ G726-32 codec, a configuration parameter called 'g726nonstandard' has been added
+ to sip.conf, so that Asterisk can use the packing order expected by the device (even
+ though it requested a different order). In addition, the internal format number for
+ G726-32 has been changed, and the old number is now assigned to AAL2-G726-32. The
+ result of this is that this version of Asterisk will be able to interoperate over
+ IAX2 with older versions of Asterisk, as long as this version is told to allow
+ 'g726aal2' instead of 'g726' as the codec for the call.
+* 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
+ 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:
+ ASTLIBDIR /usr/local/lib/asterisk
+ ASTVARLIBDIR /usr/local/share/asterisk
+ ASTETCDIR /usr/local/etc/asterisk
+ ASTBINDIR /usr/local/bin/asterisk
+ ASTSBINDIR /usr/local/sbin/asterisk
+Music on Hold:
+* The music on hold handling has been changed in some significant ways in hopes
+ to make it work in a way that is much less confusing to users. Behavior will
+ not change if the same configuration is used from older versions of Asterisk.
+ However, there are some new configuration options that will make things work
+ in a way that makes more sense.
+ Previously, many of the channel drivers had an option called "musicclass" or
+ something similar. This option set what music on hold class this channel
+ would *hear* when put on hold. Some people expected (with good reason) that
+ this option was to configure what music on hold class to play when putting
+ the bridged channel on hold. This option has now been deprecated.
+ Two new music on hold related configuration options for channel drivers have
+ been introduced. Some channel drivers support both options, some just one,
+ and some support neither of them. Check the sample configuration files to see
+ which options apply to which channel driver.
+ The "mohsuggest" option specifies which music on hold class to suggest to the
+ bridged channel when putting them on hold. The only way that this class can
+ be overridden is if the bridged channel has a specific music class set that
+ was done in the dialplan using Set(CHANNEL(musicclass)=something).
+ The "mohinterpret" option is similar to the old "musicclass" option. It
+ specifies which music on hold class this channel would like to listen to when
+ put on hold. This music class is only effective if this channel has no music
+ class set on it from the dialplan and the bridged channel putting this one on
+ hold had no "mohsuggest" setting.
+ The IAX2 and Zap channel drivers have an additional feature for the
+ "mohinterpret" option. If this option is set to "passthrough", then these
+ channel drivers will pass through the HOLD message in signalling instead of
+ starting music on hold on the channel. An example for how this would be
+ useful is in an enterprise network of Asterisk servers. When one phone on one
+ server puts a phone on a different server on hold, the remote server will be
+ responsible for playing the hold music to its local phone that was put on
+ hold instead of the far end server across the network playing the music.
+CDR Records:
+* The behavior of the "clid" field of the CDR has always been that it will
+ contain the callerid ANI if it is set, or the callerid number if ANI was not
+ set. When using the "callerid" option for various channel drivers, some
+ would set ANI and some would not. This has been cleared up so that all
+ channel drivers set ANI. If you would like to change the callerid number
+ on the channel from the dialplan and have that change also show up in the
+ CDR, then you *must* set CALLERID(ANI) as well as CALLERID(num).
+* There are some API functions that were not previously prefixed with the 'ast_'
+ prefix but now are; these include the ADSI, ODBC and AGI interfaces. If you
+ have a module that uses the services provided by res_adsi, res_odbc, or
+ res_agi, you will need to add ast_ prefixes to the functions that you call
+ from those modules.
+* format_wav: The GAIN preprocessor definition has been changed from 2 to 0
+ in Asterisk 1.4. This change was made in response to user complaints of
+ choppiness or the clipping of loud signal peaks. The GAIN preprocessor
+ definition will be retained in Asterisk 1.4, but will be removed in a
+ future release. The use of GAIN for the increasing of voicemail message
+ volume should use the 'volgain' option in voicemail.conf