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(parent: 2c3bc8b) | patch
Start out with cleared RTP payload structures instead of defaults. This should preven...
authorJoshua Colp <jcolp@digium.com>
Mon, 24 Jul 2006 15:47:59 +0000 (15:47 +0000)
committerJoshua Colp <jcolp@digium.com>
Mon, 24 Jul 2006 15:47:59 +0000 (15:47 +0000)
commitcbf79ca4890dd0faabb4fbbbe927e389dbc241d3
tree05d26d07d8906297fc5d2dd0a8115d627c47ecaftree | snapshot
parent2c3bc8b1b342ebea4e34f284e89cc99a9d84e94ccommit | diff
Start out with cleared RTP payload structures instead of defaults. This should prevent issues where if a stream (audio/stream) is not present and it's RTP payload structure is combined with the overall capability then the capability would be every codec that Asterisk supports.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Mirror of https://github.com/asterisk/asterisk.git