219 lines
9.1 KiB
Plaintext
219 lines
9.1 KiB
Plaintext
=========================================================
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===
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=== Information for upgrading from Asterisk 1.0 to 1.2
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===
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=== This file documents all the changes that MUST be taken
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=== into account when upgrading between the Asterisk
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=== versions listed below. These changes may require that
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=== you modify your configuration files, dialplan or (in
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=== some cases) source code if you have your own Asterisk
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=== modules or patches. These files also includes advance
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=== notice of any functionality that has been marked as
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=== 'deprecated' and may be removed in a future release,
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=== along with the suggested replacement functionality.
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===
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=========================================================
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Compiling:
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* The Asterisk 1.2 source code now uses C language features
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supported only by 'modern' C compilers. Generally, this means GCC
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version 3.0 or higher, although some GCC 2.96 releases will also
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work. Some non-GCC compilers that support C99 and the common GCC
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extensions (including anonymous structures and unions) will also
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work. All releases of GCC 2.95 do _not_ have the requisite feature
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support; systems using that compiler will need to be upgraded to
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a more recent compiler release.
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Dialplan Expressions:
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* The dialplan expression parser (which handles $[ ... ] constructs)
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has gone through a major upgrade, but has one incompatible change:
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spaces are no longer required around expression operators, including
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string comparisons. However, you can now use quoting to keep strings
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together for comparison. For more details, please read the
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doc/README.variables file, and check over your dialplan for possible
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problems.
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Agents:
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* The default for ackcall has been changed to "no" instead of "yes"
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because of a bug which caused the "yes" behavior to generally act like
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"no". You may need to adjust the value if your agents behave
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differently than you expect with respect to acknowledgement.
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* The AgentCallBackLogin application now requires a second '|' before
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specifying an extension@context. This is to distinguish the options
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string from the extension, so that they do not conflict. See
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'show application AgentCallbackLogin' for more details.
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Parking:
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* Parking behavior has changed slightly; when a parked call times out,
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Asterisk will attempt to deliver the call back to the extension that
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parked it, rather than the 's' extension. If that extension is busy
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or unavailable, the parked call will be lost.
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Dialing:
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* The Caller*ID of the outbound leg is now the extension that was
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called, rather than the Caller*ID of the inbound leg of the call. The
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"o" flag for Dial can be used to restore the original behavior if
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desired. Note that if you are looking for the originating callerid
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from the manager event, there is a new manager event "Dial" which
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provides the source and destination channels and callerid.
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IAX:
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* The naming convention for IAX channels has changed in two ways:
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1. The call number follows a "-" rather than a "/" character.
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2. The name of the channel has been simplified to IAX2/peer-callno,
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rather than IAX2/peer@peer-callno or even IAX2/peer@peer/callno.
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SIP:
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* The global option "port" in 1.0.X that is used to set which port to
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bind to has been changed to "bindport" to be more consistent with
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the other channel drivers and to avoid confusion with the "port"
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option for users/peers.
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* The "Registry" event now uses "Username" rather than "User" for
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consistency with IAX.
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Applications:
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* With the addition of dialplan functions (which operate similarly
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to variables), the SetVar application has been renamed to Set.
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* The CallerPres application has been removed. Use SetCallerPres
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instead. It accepts both numeric and symbolic names.
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* The applications GetGroupCount, GetGroupMatchCount, SetGroup, and
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CheckGroup have been deprecated in favor of functions. Here is a
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table of their replacements:
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GetGroupCount([groupname][@category] GROUP_COUNT([groupname][@category]) Set(GROUPCOUNT=${GROUP_COUNT()})
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GroupMatchCount(groupmatch[@category]) GROUP_MATCH_COUNT(groupmatch[@category]) Set(GROUPCOUNT=${GROUP_MATCH_COUNT(SIP/.*)})
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SetGroup(groupname[@category]) GROUP([category])=groupname Set(GROUP()=test)
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CheckGroup(max[@category]) N/A GotoIf($[ ${GROUP_COUNT()} > 5 ]?103)
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Note that CheckGroup does not have a direct replacement. There is
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also a new function called GROUP_LIST() which will return a space
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separated list of all of the groups set on a channel. The GROUP()
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function can also return the name of the group set on a channel when
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used in a read environment.
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* The applications DBGet and DBPut have been deprecated in favor of
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functions. Here is a table of their replacements:
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DBGet(foo=family/key) Set(foo=${DB(family/key)})
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DBPut(family/key=${foo}) Set(DB(family/key)=${foo})
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* The application SetLanguage has been deprecated in favor of the
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function LANGUAGE().
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SetLanguage(fr) Set(LANGUAGE()=fr)
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The LANGUAGE function can also return the currently set language:
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Set(MYLANG=${LANGUAGE()})
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* The applications AbsoluteTimeout, DigitTimeout, and ResponseTimeout
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have been deprecated in favor of the function TIMEOUT(timeouttype):
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AbsoluteTimeout(300) Set(TIMEOUT(absolute)=300)
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DigitTimeout(15) Set(TIMEOUT(digit)=15)
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ResponseTimeout(15) Set(TIMEOUT(response)=15)
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The TIMEOUT() function can also return the currently set timeouts:
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Set(DTIMEOUT=${TIMEOUT(digit)})
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* The applications SetCIDName, SetCIDNum, and SetRDNIS have been
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deprecated in favor of the CALLERID(datatype) function:
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SetCIDName(Joe Cool) Set(CALLERID(name)=Joe Cool)
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SetCIDNum(2025551212) Set(CALLERID(number)=2025551212)
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SetRDNIS(2024561414) Set(CALLERID(RDNIS)=2024561414)
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* The application Record now uses the period to separate the filename
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from the format, rather than the colon.
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* The application VoiceMail now supports a 'temporary' greeting for each
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mailbox. This greeting can be recorded by using option 4 in the
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'mailbox options' menu, and 'change your password' option has been
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moved to option 5.
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* The application VoiceMailMain now only matches the 'default' context if
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none is specified in the arguments. (This was the previously
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documented behavior, however, we didn't follow that behavior.) The old
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behavior can be restored by setting searchcontexts=yes in voicemail.conf.
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Queues:
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* A queue is now considered empty not only if there are no members but if
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none of the members are available (e.g. agents not logged on). To
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restore the original behavior, use "leavewhenempty=strict" or
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"joinwhenempty=strict" instead of "=yes" for those options.
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* It is now possible to use multi-digit extensions in the exit context
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for a queue (although you should not have overlapping extensions,
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as there is no digit timeout). This means that the EXITWITHKEY event
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in queue_log can now contain a key field with more than a single
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character in it.
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Extensions:
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* By default, there is a new option called "autofallthrough" in
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extensions.conf that is set to yes. Asterisk 1.0 (and earlier)
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behavior was to wait for an extension to be dialed after there were no
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more extensions to execute. "autofallthrough" changes this behavior
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so that the call will immediately be terminated with BUSY,
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CONGESTION, or HANGUP based on Asterisk's best guess. If you are
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writing an extension for IVR, you must use the WaitExten application
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if "autofallthrough" is set to yes.
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AGI:
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* AGI scripts did not always get SIGHUP at the end, previously. That
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behavior has been fixed. If you do not want your script to terminate
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at the end of AGI being called (e.g. on a hangup) then set SIGHUP to
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be ignored within your application.
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* CallerID is reported with agi_callerid and agi_calleridname instead
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of a single parameter holding both.
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Music On Hold:
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* The preferred format for musiconhold.conf has changed; please see the
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sample configuration file for the new format. The existing format
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is still supported but will generate warnings when the module is loaded.
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chan_modem:
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* All the chan_modem channel drivers (aopen, bestdata and i4l) are deprecated
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in this release, and will be removed in the next major Asterisk release.
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Please migrate to chan_misdn for ISDN interfaces; there is no upgrade
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path for aopen and bestdata modem users.
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MeetMe:
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* The conference application now allows users to increase/decrease their
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speaking volume and listening volume (independently of each other and
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other users); the 'admin' and 'user' menus have changed, and new sound
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files are included with this release. However, if a user calling in
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over a Zaptel channel that does NOT have hardware DTMF detection
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increases their speaking volume, it is likely they will no longer be
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able to enter/exit the menu or make any further adjustments, as the
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software DTMF detector will not be able to recognize the DTMF coming
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from their device.
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GetVar Manager Action:
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* Previously, the behavior of the GetVar manager action reported the value
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of a variable in the following manner:
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> name: value
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This has been changed to a manner similar to the SetVar action and is now
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> Variable: name
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> Value: value
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