2005-11-01 19:57:20 +00:00
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Information for Upgrading From Previous Asterisk Releases
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=========================================================
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2005-02-14 23:20:01 +00:00
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2005-03-09 06:42:56 +00:00
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Applications:
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2005-12-21 08:44:51 +00:00
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* In previous Asterisk releases, many applications would jump to priority n+101
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to indicate some kind of status or error condition. This functionality was
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marked deprecated in Asterisk 1.2. An option to disable it was provided with
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the default value set to 'on'. The default value for the global priority
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jumping option is now 'off'.
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2005-12-26 16:48:12 +00:00
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* The applications Cut, Sort, DBGet, DBPut, SetCIDNum, SetCIDName, SetRDNIS,
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AbsoluteTimeout, DigitTimeout, ResponseTimeout, SetLanguage, GetGroupCount,
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and GetGroupMatchCount were all deprecated in version 1.2, and therefore have
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been removed in this version. You should use the equivalent dialplan
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function in places where you have previously used one of these applications.
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* The application SetVar has been renamed to Set. The syntax SetVar was marked
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deprecated in version 1.2 and is no longer recognized in this version.
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2006-01-05 19:55:03 +00:00
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* app_read has been updated to use the newer options codes, using "skip" or
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"noanswer" will not work. Use s or n. Also there is a new feature i, for
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using indication tones, so typing in skip would give you unexpected results.
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2006-01-30 06:07:05 +00:00
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2005-12-26 16:48:12 +00:00
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Variables:
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* The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
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${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE},
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and ${LANGUAGE} have all been deprecated in favor of their related dialplan
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functions. You are encouraged to move towards the associated dialplan
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function, as these variables will be removed in a future release.
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2006-01-30 06:07:05 +00:00
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Functions:
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* The function ${CHECK_MD5()} has been deprecated in favor of using an
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expression: $[${MD5(<string>)} = ${saved_md5}].
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2006-01-04 09:10:56 +00:00
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The SIP channel:
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* The "incominglimit" setting is replaced by the "call-limit" setting in sip.conf.
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