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diff --git a/1.4.23-rc4/UPGRADE.txt b/1.4.23-rc4/UPGRADE.txt deleted file mode 100644 index 6dad890ea..000000000 --- a/1.4.23-rc4/UPGRADE.txt +++ /dev/null @@ -1,500 +0,0 @@ -========================================================= -=== Information for upgrading from Asterisk 1.2 to 1.4 -=== -=== -=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2 -=== UPGRADE.txt -- Upgrade info for 1.2 to 1.4 -========================================================= - -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 -'/etc/asterisk.makeopts'. - -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 -installed. - -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 '!'. - -Applications: - -* 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 - "SUCCESS". - -* 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. - -Manager: - -* 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 - -Variables: - -* The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM}, - ${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE}, - 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}, - ${OSPOUTHANDLE}, ${OSPINTOKEN}, ${OSPOUTTOKEN}, ${OSPCALLING}, - ${OSPINTIMELIMIT}, and ${OSPOUTTIMELIMIT} - -* 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. - -Functions: - -* 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 - QUEUE_MEMBER_COUNT(). - -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. - -Installation: - -* 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). - -API: - -* 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. - -Formats: - -* 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 - -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. |