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Commit a760a033 authored by Russell Bryant's avatar Russell Bryant
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Merged revisions 105932 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008) | 5 lines

Fix a bug that I just noticed in the RTP code.  The calculation for setting the
len field in an ast_frame of audio was wrong when G.722 is in use.  The len field
represents the number of ms of audio that the frame contains.  It would have
set the value to be twice what it should be.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
parent 39b9f810
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