~singpolyma/asterisk

ref: upstream/16.2.1_dfsg asterisk/UPGRADE-14.txt -rw-r--r-- 5.0 KiB
fe6d69c0Stephen Paul Weber import-tar 6 months ago
                                                                                
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===========================================================
===
=== Information for upgrading between Asterisk versions
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also include advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
=== UPGRADE-10.txt  -- Upgrade info for 1.8 to 10
=== UPGRADE-11.txt  -- Upgrade info for 10 to 11
=== UPGRADE-12.txt  -- Upgrade info for 11 to 12
=== UPGRADE-13.txt  -- Upgrade info for 12 to 13
===========================================================

From 14.6.0 to 14.7.0:

Core:
 - ast_app_parse_timelen now returns an error if it encounters extra characters
   at the end of the string to be parsed.

From 14.4.0 to 14.5.0:

Core:
 - Support for embedded modules has been removed.  This has not worked in
   many years.  LOADABLE_MODULES menuselect option is also removed as
   loadable module support is now always enabled.

From 14.3.0 to 14.4.0:

res_rtp_asterisk:
 - The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
   Data and Control Packets on a Single Port." For the PJSIP channel driver,
   chan_pjsip, you can set "rtcp_mux = yes" on a PJSIP endpoint in pjsip.conf
   to enable the feature. For chan_sip you can set "rtcp_mux = yes" either
   globally or on a per-peer basis in sip.conf.

New in 14.0.0

ARI:
 - The policy for when to send "Dial" events has changed. Previously, "Dial"
   events were sent on the calling channel's topic. However, starting in Asterisk
   14, if there is no calling channel on which to send the event, the event is
   instead sent on the called channel's topic. Note that for the ARI channels
   resource's dial operation, this means that the "Dial" events will always be
   sent on the called channel's topic.

Channel Drivers:

chan_dahdi:
 - For users using the FXO port (FXS signaling) distinctive ring detection
   feature, you will need to adjust the dringX count values.  The count
   values now only record ring end events instead of any DAHDI event.  A
   ring-ring-ring pattern would exceed the pattern limits and stop
   Caller-ID detection.

chan_sip:
 - The SIP dial string has been extended past the [!dnid] option by another
   exclamation mark: [!dnid[!fromuri].  An exclamation mark in the To-URI
   will now mean changes to the From-URI.

Core:
 - The REF_DEBUG compiler flag is now used to enable refdebug by default.
   The setting can be overridden in asterisk.conf by setting refdebug in
   the options category.  No recompile is required to enable/disable it.

 - Modified processing of command-line options to first parse only what
   is necessary to read asterisk.conf. Once asterisk.conf is fully loaded,
   the remaining options are processed.  The -X option now applies to
   asterisk.conf only.  To enable #exec for other config files you must
   set execincludes=yes in asterisk.conf.  Any other option set on the
   command-line will now override the equivalent setting from asterisk.conf.

AMI:
 - The 'ModuleCheck' Action's Version key will no longer show the module
   version. The value will always be blank.

CLI:
 - The 'core show file version' command has been removed. When Asterisk
   moved to Git, the source control version support was removed. As a
   result, the CLi command was no longer useful and was removed as well.

Logging:
 - The first callid created is now 1 instead of 0.  The value 0
   is now reserved to represent a lack of callid.

AMI:
 - The Command action now sends the output from the CLI command as a series
   of Output headers for each line instead of as a block of text with the
   --END COMMAND-- delimiter to match the output from other actions.

   Commands that fail to execute (no such command, invalid syntax etc.) now
   return an Error response instead of Success.

app_amd:
 - The 'maximum_number_of_words' configuration option and parameter to the AMD
   application previously did not match the documented functionality + variable
   name.  In Asterisk 13, a value of '3' would mean that if '3' words were detected,
   the result would be detection as a 'MACHINE'.  As of this version, the value
   reflects the maximum words that if EXCEEDED (rather than reached), would
   result in detection as a machine.  This means that you should update this
   value to be one higher than your previos value, if your previous value
   was working well for you.

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