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author | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-06-27 09:51:45 +0000 |
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committer | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-06-27 09:51:45 +0000 |
commit | 33ecd470c0d85b05e23fe616fff6159712628a1d (patch) | |
tree | 51fc68901dbcd7ff66a9def48ce58cb13f519a2f /UPGRADE.txt | |
parent | b59b9b3041b30c20f4598725a857dfd59afa426e (diff) |
Minor tweaks and spelling fixes for CHANGES and UPGRADE.txt.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203960 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'UPGRADE.txt')
-rw-r--r-- | UPGRADE.txt | 5 |
1 files changed, 5 insertions, 0 deletions
diff --git a/UPGRADE.txt b/UPGRADE.txt index 6272b9948..d670e414d 100644 --- a/UPGRADE.txt +++ b/UPGRADE.txt @@ -20,11 +20,16 @@ From 1.6.2 to 1.6.3: +* The rarely used 'event_log' and LOG_EVENT channel have been removed; the few + users of this channel in the tree have been converted to LOG_NOTICE or removed + (in cases where the same message was already generated to another channel). + * The usage of RTP inside of Asterisk has now become modularized. This means the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk. If you are not using autoload=yes in modules.conf you will need to ensure it is set to load. If not, then any module which uses RTP (such as chan_sip) will not be able to send or receive calls. + * The app_dahdiscan.c file has been removed, but the dialplan app DAHDIScan still remains. It now exists within app_chanspy.c and retains the exact same functionality as before. |