From dcca8f345fa55b146fe6b0930df18b226809cc70 Mon Sep 17 00:00:00 2001 From: Russell Bryant Date: Fri, 19 Jan 2007 18:06:03 +0000 Subject: 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: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- channels/chan_alsa.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'channels/chan_alsa.c') 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; } -- cgit v1.2.3