diff options
| author | Richard Mudgett <rmudgett@digium.com> | 2017-08-09 15:24:58 -0500 |
|---|---|---|
| committer | Richard Mudgett <rmudgett@digium.com> | 2017-08-10 11:46:42 -0500 |
| commit | ecd1f87edf71158f836f81a4b2dabccc7b394726 (patch) | |
| tree | 0fd2feabc2c619354be3dadd2114317ae984c7be /UPGRADE-14.txt | |
| parent | 4ed2733ddea705761f0e340cdcba972a5f8be68a (diff) | |
UPGRADE notes: Prepare for the eventual 16 branch.
Change-Id: I4ca2f07ed62d77f1fdd10c3b216f6a28dd75720c
Diffstat (limited to 'UPGRADE-14.txt')
| -rw-r--r-- | UPGRADE-14.txt | 24 |
1 files changed, 24 insertions, 0 deletions
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 |