From ecd1f87edf71158f836f81a4b2dabccc7b394726 Mon Sep 17 00:00:00 2001 From: Richard Mudgett Date: Wed, 9 Aug 2017 15:24:58 -0500 Subject: UPGRADE notes: Prepare for the eventual 16 branch. Change-Id: I4ca2f07ed62d77f1fdd10c3b216f6a28dd75720c --- UPGRADE-14.txt | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) (limited to 'UPGRADE-14.txt') diff --git a/UPGRADE-14.txt b/UPGRADE-14.txt index f8fa7906b..aaf236ba2 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 -- cgit v1.2.3