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diff --git a/UPGRADE.txt b/UPGRADE.txt new file mode 100644 index 000000000..6dad890ea --- /dev/null +++ b/UPGRADE.txt @@ -0,0 +1,500 @@ +========================================================= +=== 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. |