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authorJoshua Colp <jcolp@digium.com>2006-07-24 15:47:59 +0000
committerJoshua Colp <jcolp@digium.com>2006-07-24 15:47:59 +0000
commitcbf79ca4890dd0faabb4fbbbe927e389dbc241d3 (patch)
tree05d26d07d8906297fc5d2dd0a8115d627c47ecaf /codecs/gsm/Makefile
parent2c3bc8b1b342ebea4e34f284e89cc99a9d84e94c (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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