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117 lines
5.8 KiB
117 lines
5.8 KiB
===========================================================
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===
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=== Information for upgrading between Asterisk 1.6 versions
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===
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=== These files document 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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=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
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=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
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=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
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===
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===========================================================
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From 1.6.1 to 1.6.2:
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* The res_indications module has been removed. Its functionality was important
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enough that most of it has been moved into the Asterisk core.
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Two applications previously provided by res_indications, PlayTones and
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StopPlayTones, have been moved into a new module, app_playtones.
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From 1.6.0.1 to 1.6.1:
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* The ast_agi_register_multiple() and ast_agi_unregister_multiple()
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API calls were added in 1.6.0, so that modules that provide multiple
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AGI commands could register/unregister them all with a single
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step. However, these API calls were not implemented properly, and did
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not allow the caller to know whether registration or unregistration
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succeeded or failed. They have been redefined to now return success
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or failure, but this means any code using these functions will need
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be recompiled after upgrading to a version of Asterisk containing
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these changes. In addition, the source code using these functions
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should be reviewed to ensure it can properly react to failure
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of registration or unregistration of its API commands.
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* The ast_agi_fdprintf() API call has been renamed to ast_agi_send()
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to better match what it really does, and the argument order has been
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changed to be consistent with other API calls that perform similar
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operations.
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From 1.6.0.x to 1.6.1:
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* In previous versions of Asterisk, due to the way objects were arranged in
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memory by chan_sip, the order of entries in sip.conf could be adjusted to
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control the behavior of matching against peers and users. The way objects
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are managed has been significantly changed for reasons involving performance
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and stability. A side effect of these changes is that the order of entries
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in sip.conf can no longer be relied upon to control behavior.
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* The following core commands dealing with dialplan have been deprecated: 'core
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show globals', 'core set global' and 'core set chanvar'. Use the equivalent
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'dialplan show globals', 'dialplan set global' and 'dialplan set chanvar'
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instead.
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* In the dialplan expression parser, the logical value of spaces
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immediately preceding a standalone 0 previously evaluated to
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true. It now evaluates to false. This has confused a good many
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people in the past (typically because they failed to realize the
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space had any significance). Since this violates the Principle of
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Least Surprise, it has been changed.
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* While app_directory has always relied on having a voicemail.conf or users.conf file
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correctly set up, it now is dependent on app_voicemail being compiled as well.
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* SIP: All of the functionality in SIPCHANINFO() has been implemented in CHANNEL(),
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and you should start using that function instead for retrieving information about
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the channel in a technology-agnostic way.
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* If you have any third party modules which use a config file variable whose
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name ends in a '+', please note that the append capability added to this
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version may now conflict with that variable naming scheme. An easy
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workaround is to ensure that a space occurs between the '+' and the '=',
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to differentiate your variable from the append operator. This potential
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conflict is unlikely, but is documented here to be thorough.
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* The "Join" event from app_queue now uses the CallerIDNum header instead of
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the CallerID header to indicate the CallerID number.
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From 1.6.1 to 1.6.2:
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* The default console now will use colors according to the default background
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color, instead of forcing the background color to black. If you are using a
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light colored background for your console, you may wish to use the option
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flag '-W' to present better color choices for the various messages. However,
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if you'd prefer the old method of forcing colors to white text on a black
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background, the compatibility option -B is provided for this purpose.
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* SendImage() no longer hangs up the channel on transmission error or on
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any other error; in those cases, a FAILURE status is stored in
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SENDIMAGESTATUS and dialplan execution continues. The possible
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return values stored in SENDIMAGESTATUS are: SUCCESS, FAILURE, and
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UNSUPPORTED. ('OK' has been replaced with 'SUCCESS', and 'NOSUPPORT'
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has been replaced with 'UNSUPPORTED'). This change makes the
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SendImage application more consistent with other applications.
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* skinny.conf now has separate sections for lines and devices.
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Please have a look at configs/skinny.conf.sample and update
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your skinny.conf.
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* Queue names previously were treated in a case-sensitive manner,
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meaning that queues with names like "sales" and "sALeS" would be
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seen as unique queues. The parsing logic has changed to use
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case-insensitive comparisons now when originally hashing based on
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queue names, meaning that now the two queues mentioned as examples
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earlier will be seen as having the same name.
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* The SPRINTF() dialplan function has been moved into its own module,
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func_sprintf, and is no longer included in func_strings. If you use this
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function and do not use 'autoload=yes' in modules.conf, you will need
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to explicitly load func_sprintf for it to be available.
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