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Add an upgrade note for libuuid dependency; remove note in CHANGES

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This patch notes that libuuid is now a dependency for res_rtp_asterisk; this
was introduced in between 11.4.0 and 11.5.0 to resolve a dependency for
pjproject, which res_rtp_asterisk uses for ICE/STUN/TURN support.

It also removes a conflicting note from CHANGES. While support for playing
prompts to the first participant was added for app_queue, it was disabled
by default and an option added to enable it. That was properly noted in the
UPGRADE.txt file.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@395020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Matthew Jordan
2013-07-21 22:51:58 +00:00
parent 5b3d9a9b20
commit 06f0da01f5
2 changed files with 7 additions and 12 deletions
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10
CHANGES
View File

@@ -7,19 +7,11 @@
=== and the other UPGRADE files for older releases.
===
==============================================================================
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
------------------------------------------------------------------------------
App_queue
---------
* App_queue will now play periodic announcements for the caller that
holds the first position in the queue while waiting for answer.
------------------------------------------------------------------------------
--- Functionality changes since Asterisk 1.8.12.0 ----------------------------
------------------------------------------------------------------------------
Build System
-------------------
* The Asterisk build system will now build and install a shared library

9
UPGRADE.txt
View File

@@ -30,6 +30,9 @@ From 11.4 to 11.5:
default but can be enabled on an individual queue using the 'announce-to-first-user'
option.
* The libuuid development library is now required for res_rtp_asterisk. Consult
your distribution for the appropriate development library name.
From 11.3 to 11.4:
* Added the 'n' option to MeetMe to prevent application of the DENOISE function
to a channel joining a conference. Some channel drivers that vary the number
@@ -185,7 +188,7 @@ SIP
configuration option. Symptoms of this include one way media or no media flow.
chan_unistim
- Due to massive update in chan_unistim phone keys functions and on-screen
- Due to massive update in chan_unistim phone keys functions and on-screen
information changed.
users.conf:
@@ -257,10 +260,10 @@ Manager:
unchanged.
Module Support Level
- All modules in the addons, apps, bridge, cdr, cel, channels, codecs,
- All modules in the addons, apps, bridge, cdr, cel, channels, codecs,
formats, funcs, pbx, and res have been updated to include MODULEINFO data
that includes <support_level> tags with a value of core, extended, or deprecated.
More information is available on the Asterisk wiki at
More information is available on the Asterisk wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
Deprecated modules are now marked to not build by default and must be explicitly