d23b94c206
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
58 lines
2.3 KiB
Text
Executable file
58 lines
2.3 KiB
Text
Executable file
Information for Upgrading From Asterisk 1.0
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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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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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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 a minor way such
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that the call number follows a "-" rather than a "/" character.
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Applications:
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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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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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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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