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86 lines
3.5 KiB
Plaintext
86 lines
3.5 KiB
Plaintext
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===========================================================
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===
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=== Information for upgrading between Asterisk 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 include 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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=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
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=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
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=== UPGRADE-11.txt -- Upgrade info for 10 to 11
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=== UPGRADE-12.txt -- Upgrade info for 11 to 12
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=== UPGRADE-13.txt -- Upgrade info for 12 to 13
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=== UPGRADE-14.txt -- Upgrade info for 13 to 14
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=== UPGRADE-15.txt -- Upgrade info for 14 to 15
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===========================================================
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New in 15.0.0:
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Core:
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- The 'Data Retrieval API' has been removed. This API was not actively
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maintained, was not added to new modules (such as res_pjsip), and there
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exist better alternatives to acquire the same information, such as the
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ARI. As a result, the 'DataGet' AMI action as well as the 'data get'
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CLI command have been removed.
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From 14.6.0 to 14.7.0:
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Core:
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- ast_app_parse_timelen now returns an error if it encounters extra characters
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at the end of the string to be parsed.
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From 14.4.0 to 14.5.0:
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Core:
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- Support for embedded modules has been removed. This has not worked in
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many years. LOADABLE_MODULES menuselect option is also removed as
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loadable module support is now always enabled.
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From 14.3.0 to 14.4.0:
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res_rtp_asterisk:
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- The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
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Data and Control Packets on a Single Port." For the PJSIP channel driver,
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chan_pjsip, you can set "rtcp_mux = yes" on a PJSIP endpoint in pjsip.conf
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to enable the feature. For chan_sip you can set "rtcp_mux = yes" either
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globally or on a per-peer basis in sip.conf.
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New in 14.0.0
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ARI:
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- The policy for when to send "Dial" events has changed. Previously, "Dial"
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events were sent on the calling channel's topic. However, starting in Asterisk
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14, if there is no calling channel on which to send the event, the event is
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instead sent on the called channel's topic. Note that for the ARI channels
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resource's dial operation, this means that the "Dial" events will always be
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sent on the called channel's topic.
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Queue:
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- When reloading the members of a queue, the members added dynamically (i.e.
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added via the CLI command "queue add" or the AMI action "QueueAdd") now have
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their ringinuse value updated to the value of the queue. Previously, the
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ringinuse value for dynamic members was not updated on reload.
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Queue log:
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- New RINGCANCELED event is logged when the caller hangs up while ringing.
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The data1 field contains number of miliseconds since start of ringing.
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Channel Drivers:
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chan_dahdi:
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- Support for specifying a DAHDI channel using a path under /dev/dahdi
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("by name") has been removed. It was never used. Instead you should
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use kernel-level channel number allocation using span assignments.
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See the documentation of dahdi-linux and dahdi-tools.
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