[asterisk-commits] russell: branch 1.6.0 r178145 - in /branches/1.6.0: ./ main/rtp.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Feb 23 17:17:31 CST 2009
Author: russell
Date: Mon Feb 23 17:17:30 2009
New Revision: 178145
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=178145
Log:
Merged revisions 178142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009) | 22 lines
Merged revisions 178141 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines
Fix infinite DTMF when a BEGIN is received without an END.
This commit is related to rev 175124 of 1.4 where a previous attempt was made
to fix this problem. The problem with the previous patch was that the inserted
code needed to go _before_ setting the lastrxts to the current timestamp.
Because those were the same, the dtmfcount variable was never decremented, and
so the END was never sent.
In passing, I removed the dtmfsamples variable which was completed unused. I
also removed a redundant setting of the lastrxts variable.
(closes issue #14460)
Reported by: moliveras
........
................
Modified:
branches/1.6.0/ (props changed)
branches/1.6.0/main/rtp.c
Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.0/main/rtp.c
URL: http://svn.digium.com/svn-view/asterisk/branches/1.6.0/main/rtp.c?view=diff&rev=178145&r1=178144&r2=178145
==============================================================================
--- branches/1.6.0/main/rtp.c (original)
+++ branches/1.6.0/main/rtp.c Mon Feb 23 17:17:30 2009
@@ -1590,13 +1590,7 @@
rtp->lastrxformat = rtp->f.subclass = rtpPT.code;
rtp->f.frametype = (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) ? AST_FRAME_VOICE : (rtp->f.subclass & AST_FORMAT_VIDEO_MASK) ? AST_FRAME_VIDEO : AST_FRAME_TEXT;
- if (!rtp->lastrxts)
- rtp->lastrxts = timestamp;
-
rtp->rxseqno = seqno;
-
- /* Record received timestamp as last received now */
- rtp->lastrxts = timestamp;
if (rtp->dtmfcount) {
rtp->dtmfcount -= (timestamp - rtp->lastrxts);
@@ -1612,6 +1606,9 @@
return f;
}
}
+
+ /* Record received timestamp as last received now */
+ rtp->lastrxts = timestamp;
rtp->f.mallocd = 0;
rtp->f.datalen = res - hdrlen;
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