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Merged revisions 51311 via svnmerge from

Browse Source 
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
This commit is contained in:
Russell Bryant
2007-01-19 18:06:03 +00:00
parent 9d509eaf76
commit dcca8f345f
19 changed files with 223 additions and 128 deletions
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24
channels/chan_gtalk.c
View File

@@ -170,7 +170,9 @@ AST_MUTEX_DEFINE_STATIC(gtalklock); /*!< Protect the interface list (of gtalk_pv
/* Forward declarations */
static struct ast_channel *gtalk_request(const char *type, int format, void *data, int *cause);
static int gtalk_digit(struct ast_channel *ast, char digit);
static int gtalk_digit(struct ast_channel *ast, char digit, unsigned int duration);
static int gtalk_digit_begin(struct ast_channel *ast, char digit);
static int gtalk_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
static int gtalk_call(struct ast_channel *ast, char *dest, int timeout);
static int gtalk_hangup(struct ast_channel *ast);
static int gtalk_answer(struct ast_channel *ast);
@@ -195,8 +197,8 @@ static const struct ast_channel_tech gtalk_tech = {
.description = "Gtalk Channel Driver",
.capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
.requester = gtalk_request,
.send_digit_begin = gtalk_digit,
.send_digit_end = gtalk_digit,
.send_digit_begin = gtalk_digit_begin,
.send_digit_end = gtalk_digit_end,
.bridge = ast_rtp_bridge,
.call = gtalk_call,
.hangup = gtalk_hangup,
@@ -1338,7 +1340,17 @@ static int gtalk_indicate(struct ast_channel *ast, int condition, const void *da
return res;
}
static int gtalk_digit(struct ast_channel *ast, char digit)
static int gtalk_digit_begin(struct ast_channel *chan, char digit)
{
return gtalk_digit(chan, digit, 0);
}
static int gtalk_digit_end(struct ast_channel *chan, char digit, unsigned int duration)
{
return gtalk_digit(chan, digit, duration);
}
static int gtalk_digit(struct ast_channel *ast, char digit, unsigned int duration)
{
struct gtalk_pvt *p = ast->tech_pvt;
struct gtalk *client = p->parent;
@@ -1373,8 +1385,8 @@ static int gtalk_digit(struct ast_channel *ast, char digit)
iks_insert_node(gtalk, dtmf);
ast_mutex_lock(&p->lock);
if(ast->dtmff.frametype == AST_FRAME_DTMF) {
ast_verbose("Sending 250ms dtmf!\n");
if (ast->dtmff.frametype == AST_FRAME_DTMF) {
ast_log(LOG_DEBUG, "Sending 250ms dtmf!\n");
} else if (ast->dtmff.frametype == AST_FRAME_DTMF_BEGIN) {
iks_insert_attrib(dtmf, "action", "button-down");
} else if (ast->dtmff.frametype == AST_FRAME_DTMF_END) {