[Asterisk-code-review] UPGRADE notes: Prepare for the eventual 16 branch. (asterisk[master])
Jenkins2
asteriskteam at digium.com
Thu Aug 10 20:09:25 CDT 2017
Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/6208 )
Change subject: UPGRADE notes: Prepare for the eventual 16 branch.
......................................................................
UPGRADE notes: Prepare for the eventual 16 branch.
Change-Id: I4ca2f07ed62d77f1fdd10c3b216f6a28dd75720c
---
M UPGRADE-14.txt
M UPGRADE-15.txt
M UPGRADE.txt
3 files changed, 49 insertions(+), 56 deletions(-)
Approvals:
George Joseph: Looks good to me, but someone else must approve
Joshua Colp: Looks good to me, approved
Jenkins2: Approved for Submit
diff --git a/UPGRADE-14.txt b/UPGRADE-14.txt
index f8fa790..aaf236b 100644
--- a/UPGRADE-14.txt
+++ b/UPGRADE-14.txt
@@ -22,6 +22,30 @@
=== 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
diff --git a/UPGRADE-15.txt b/UPGRADE-15.txt
index 2e5eb1c..30dc5d0 100644
--- a/UPGRADE-15.txt
+++ b/UPGRADE-15.txt
@@ -21,69 +21,20 @@
=== 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.
Build System:
- '--with-pjproject-bundled' is now the default when running ./configure
It can be disabled with '--without-pjproject-bundled'.
-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.
+ - Multi-stream support has been added so a channel can have multiple
+ streams of the same type such as audio and video.
-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.
-
+ - 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.
diff --git a/UPGRADE.txt b/UPGRADE.txt
index 87eabde..1fe82c0 100644
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -23,3 +23,21 @@
=== UPGRADE-14.txt -- Upgrade info for 13 to 14
=== UPGRADE-15.txt -- Upgrade info for 14 to 15
===========================================================
+
+New in 16.0.0:
+
+New in 15.0.0:
+
+Build System:
+ - '--with-pjproject-bundled' is now the default when running ./configure
+ It can be disabled with '--without-pjproject-bundled'.
+
+Core:
+ - Multi-stream support has been added so a channel can have multiple
+ streams of the same type such as audio and video.
+
+ - 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.
--
To view, visit https://gerrit.asterisk.org/6208
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-MessageType: merged
Gerrit-Change-Id: I4ca2f07ed62d77f1fdd10c3b216f6a28dd75720c
Gerrit-Change-Number: 6208
Gerrit-PatchSet: 2
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
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