diff options
| author | David Vossel <dvossel@digium.com> | 2009-03-24 20:01:29 +0000 |
|---|---|---|
| committer | David Vossel <dvossel@digium.com> | 2009-03-24 20:01:29 +0000 |
| commit | da2230adf081d932294ffe5f2629ad66cc338917 (patch) | |
| tree | 76ae843f10f9e72bf82058904748ffcab3e15aa2 /CHANGES | |
| parent | 3fd19b3ab64059865875a5bd0f5c22563f7794f4 (diff) | |
SIP preferred codec only feature
Added an option to respond to a SIP invite with only the single most preferred joint codec. This limits the options of what codecs the other side can use.
(closes issue #12485)
Reported by: bamby
Review: http://reviewboard.digium.com/r/206/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'CHANGES')
| -rw-r--r-- | CHANGES | 5 |
1 files changed, 5 insertions, 0 deletions
@@ -12,6 +12,11 @@ --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.6.3 ------------- ------------------------------------------------------------------------------ +SIP Changes +----------- + * Added preferred_codec_only option in sip.conf. This feature limits the joint + codecs sent in response to an INVITE to the single most preferred codec. + Applications ------------ * Added progress option to the app_dial D() option. When progress DTMF is |