diff options
| author | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-01-19 18:06:03 +0000 |
|---|---|---|
| committer | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-01-19 18:06:03 +0000 |
| commit | f91595d103f5afcf8b525f8481ef0bd6f9e5ac6f (patch) | |
| tree | 6fb863ed2b3e4abdc44db77e4e5cce3bed75c618 /channels/chan_alsa.c | |
| parent | 8358f66e731bcbab6d670c8cbc338fbd4103bf52 (diff) | |
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...)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels/chan_alsa.c')
| -rw-r--r-- | channels/chan_alsa.c | 7 |
1 files changed, 4 insertions, 3 deletions
diff --git a/channels/chan_alsa.c b/channels/chan_alsa.c index 6c52b3c9f..67c0b66d6 100644 --- a/channels/chan_alsa.c +++ b/channels/chan_alsa.c @@ -185,7 +185,7 @@ static int nosound = 0; /* ZZ */ static struct ast_channel *alsa_request(const char *type, int format, void *data, int *cause); -static int alsa_digit(struct ast_channel *c, char digit); +static int alsa_digit(struct ast_channel *c, char digit, unsigned int duration); static int alsa_text(struct ast_channel *c, const char *text); static int alsa_hangup(struct ast_channel *c); static int alsa_answer(struct ast_channel *c); @@ -494,10 +494,11 @@ static int soundcard_init(void) return readdev; } -static int alsa_digit(struct ast_channel *c, char digit) +static int alsa_digit(struct ast_channel *c, char digit, unsigned int duration) { ast_mutex_lock(&alsalock); - ast_verbose(" << Console Received digit %c >> \n", digit); + ast_verbose(" << Console Received digit %c of duration %u ms >> \n", + digit, duration); ast_mutex_unlock(&alsalock); return 0; } |