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: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 1 insertions, 1 deletions

diff --git a/channels/chan_misdn.c b/channels/chan_misdn.c
index 5a2c8a420..268dee1f5 100644
--- a/channels/chan_misdn.c
+++ b/channels/chan_misdn.c
@@ -2067,7 +2067,7 @@ static int misdn_digit_begin(struct ast_channel *chan, char digit)
 	return 0;
 }
 
-static int misdn_digit_end(struct ast_channel *ast, char digit )
+static int misdn_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
 {
 	struct chan_list *p;