Merged revisions 51311 via svnmerge from

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

........
r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines

Merge the changes from the /team/group/vldtmf_fixup branch.

The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged.  So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio.  However,
since there was no audio coming in, the DTMF_END was never generated.  This
caused DTMF based features to no longer work.

To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf).  If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.

Channel drivers also now get passed the length of the digit to their digit_end
callback.  This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.

(issue #8597, maybe others...)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3

main/frame.c

@@ -501,11 +501,9 @@ struct ast_frame *ast_frdup(const struct ast_frame *f)
 		strcpy((char *)out->src, f->src);
 	}
 	out->has_timing_info = f->has_timing_info;
-	if (f->has_timing_info) {
-		out->ts = f->ts;
-		out->len = f->len;
-		out->seqno = f->seqno;
-	}
+	out->ts = f->ts;
+	out->len = f->len;
+	out->seqno = f->seqno;
 	return out;
 }