<p>Richard Mudgett has uploaded this change for <strong>review</strong>.</p><p><a href="https://gerrit.asterisk.org/6208">View Change</a></p><pre style="font-family: monospace,monospace; white-space: pre-wrap;">Prepare UPGRADE notes for the eventual 16 branch.<br><br>Change-Id: I4ca2f07ed62d77f1fdd10c3b216f6a28dd75720c<br>---<br>M UPGRADE-14.txt<br>M UPGRADE-15.txt<br>M UPGRADE.txt<br>3 files changed, 41 insertions(+), 52 deletions(-)<br><br></pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/08/6208/1</pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">diff --git a/UPGRADE-14.txt b/UPGRADE-14.txt<br>index f8fa790..aaf236b 100644<br>--- a/UPGRADE-14.txt<br>+++ b/UPGRADE-14.txt<br>@@ -22,6 +22,30 @@<br> === UPGRADE-13.txt -- Upgrade info for 12 to 13<br> ===========================================================<br> <br>+From 14.6.0 to 14.7.0:<br>+<br>+Core:<br>+ - ast_app_parse_timelen now returns an error if it encounters extra characters<br>+ at the end of the string to be parsed.<br>+<br>+From 14.4.0 to 14.5.0:<br>+<br>+Core:<br>+ - Support for embedded modules has been removed. This has not worked in<br>+ many years. LOADABLE_MODULES menuselect option is also removed as<br>+ loadable module support is now always enabled.<br>+<br>+From 14.3.0 to 14.4.0:<br>+<br>+res_rtp_asterisk:<br>+ - The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP<br>+ Data and Control Packets on a Single Port." For the PJSIP channel driver,<br>+ chan_pjsip, you can set "rtcp_mux = yes" on a PJSIP endpoint in pjsip.conf<br>+ to enable the feature. For chan_sip you can set "rtcp_mux = yes" either<br>+ globally or on a per-peer basis in sip.conf.<br>+<br>+New in 14.0.0<br>+<br> ARI:<br> - The policy for when to send "Dial" events has changed. Previously, "Dial"<br> events were sent on the calling channel's topic. However, starting in Asterisk<br>diff --git a/UPGRADE-15.txt b/UPGRADE-15.txt<br>index 4ebe400..4226785 100644<br>--- a/UPGRADE-15.txt<br>+++ b/UPGRADE-15.txt<br>@@ -21,65 +21,16 @@<br> === UPGRADE-12.txt -- Upgrade info for 11 to 12<br> === UPGRADE-13.txt -- Upgrade info for 12 to 13<br> === UPGRADE-14.txt -- Upgrade info for 13 to 14<br>-=== UPGRADE-15.txt -- Upgrade info for 14 to 15<br> ===========================================================<br> <br> New in 15.0.0:<br> <br> Core:<br>+ - Multi-stream support has been added so a channel can have multiple<br>+ streams of the same type such as audio and video.<br>+<br> - The 'Data Retrieval API' has been removed. This API was not actively<br> maintained, was not added to new modules (such as res_pjsip), and there<br> exist better alternatives to acquire the same information, such as the<br> ARI. As a result, the 'DataGet' AMI action as well as the 'data get'<br> CLI command have been removed.<br>-<br>-From 14.6.0 to 14.7.0:<br>-<br>-Core:<br>- - ast_app_parse_timelen now returns an error if it encounters extra characters<br>- at the end of the string to be parsed.<br>-<br>-From 14.4.0 to 14.5.0:<br>-<br>-Core:<br>- - Support for embedded modules has been removed. This has not worked in<br>- many years. LOADABLE_MODULES menuselect option is also removed as<br>- loadable module support is now always enabled.<br>-<br>-From 14.3.0 to 14.4.0:<br>-<br>-res_rtp_asterisk:<br>- - The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP<br>- Data and Control Packets on a Single Port." For the PJSIP channel driver,<br>- chan_pjsip, you can set "rtcp_mux = yes" on a PJSIP endpoint in pjsip.conf<br>- to enable the feature. For chan_sip you can set "rtcp_mux = yes" either<br>- globally or on a per-peer basis in sip.conf.<br>-<br>-New in 14.0.0<br>-<br>-ARI:<br>- - The policy for when to send "Dial" events has changed. Previously, "Dial"<br>- events were sent on the calling channel's topic. However, starting in Asterisk<br>- 14, if there is no calling channel on which to send the event, the event is<br>- instead sent on the called channel's topic. Note that for the ARI channels<br>- resource's dial operation, this means that the "Dial" events will always be<br>- sent on the called channel's topic.<br>-<br>-Queue:<br>- - When reloading the members of a queue, the members added dynamically (i.e.<br>- added via the CLI command "queue add" or the AMI action "QueueAdd") now have<br>- their ringinuse value updated to the value of the queue. Previously, the<br>- ringinuse value for dynamic members was not updated on reload.<br>-<br>-Queue log:<br>- - New RINGCANCELED event is logged when the caller hangs up while ringing.<br>- The data1 field contains number of miliseconds since start of ringing.<br>-<br>-Channel Drivers:<br>-<br>-chan_dahdi:<br>- - Support for specifying a DAHDI channel using a path under /dev/dahdi<br>- ("by name") has been removed. It was never used. Instead you should<br>- use kernel-level channel number allocation using span assignments.<br>- See the documentation of dahdi-linux and dahdi-tools.<br>-<br>diff --git a/UPGRADE.txt b/UPGRADE.txt<br>index 87eabde..c7ca00a 100644<br>--- a/UPGRADE.txt<br>+++ b/UPGRADE.txt<br>@@ -23,3 +23,17 @@<br> === UPGRADE-14.txt -- Upgrade info for 13 to 14<br> === UPGRADE-15.txt -- Upgrade info for 14 to 15<br> ===========================================================<br>+<br>+New in 16.0.0:<br>+<br>+New in 15.0.0:<br>+<br>+Core:<br>+ - Multi-stream support has been added so a channel can have multiple<br>+ streams of the same type such as audio and video.<br>+<br>+ - The 'Data Retrieval API' has been removed. This API was not actively<br>+ maintained, was not added to new modules (such as res_pjsip), and there<br>+ exist better alternatives to acquire the same information, such as the<br>+ ARI. As a result, the 'DataGet' AMI action as well as the 'data get'<br>+ CLI command have been removed.<br></pre><p>To view, visit <a href="https://gerrit.asterisk.org/6208">change 6208</a>. To unsubscribe, visit <a href="https://gerrit.asterisk.org/settings">settings</a>.</p><div itemscope itemtype="http://schema.org/EmailMessage"><div itemscope itemprop="action" itemtype="http://schema.org/ViewAction"><link itemprop="url" href="https://gerrit.asterisk.org/6208"/><meta itemprop="name" content="View Change"/></div></div>
<div style="display:none"> Gerrit-Project: asterisk </div>
<div style="display:none"> Gerrit-Branch: master </div>
<div style="display:none"> Gerrit-MessageType: newchange </div>
<div style="display:none"> Gerrit-Change-Id: I4ca2f07ed62d77f1fdd10c3b216f6a28dd75720c </div>
<div style="display:none"> Gerrit-Change-Number: 6208 </div>
<div style="display:none"> Gerrit-PatchSet: 1 </div>
<div style="display:none"> Gerrit-Owner: Richard Mudgett <rmudgett@digium.com> </div>