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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
=== UPGRADE-14.txt -- Upgrade info for 13 to 14
=== UPGRADE-15.txt -- Upgrade info for 14 to 15
===========================================================
New in 15.0.0:
Core:
- The 'Data Retrieval API' has been removed. This API was not actively
maintained, was not added to new modules (such as res_pjsip), and there
exist better alternatives to acquire the same information, such as the
ARI. As a result, the 'DataGet' AMI action as well as the 'data get'
CLI command have been removed.
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.
Queue:
- When reloading the members of a queue, the members added dynamically (i.e.
added via the CLI command "queue add" or the AMI action "QueueAdd") now have
their ringinuse value updated to the value of the queue. Previously, the
ringinuse value for dynamic members was not updated on reload.
Queue log:
- New RINGCANCELED event is logged when the caller hangs up while ringing.
The data1 field contains number of miliseconds since start of ringing.
Channel Drivers:
chan_dahdi:
- Support for specifying a DAHDI channel using a path under /dev/dahdi
("by name") has been removed. It was never used. Instead you should
use kernel-level channel number allocation using span assignments.
See the documentation of dahdi-linux and dahdi-tools.
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