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Add TALK_DETECTED variable (bug #2628)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2004-11-01 02:04:05 +00:00
parent 219acbcc07
commit 37b375c21a
2 changed files with 7 additions and 1 deletions
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6
apps/app_talkdetect.c
View File

@@ -145,7 +145,13 @@ static int background_detect_exec(struct ast_channel *chan, void *data)
if (ms < 0)
ms = 0;
if ((ms > min) && ((max < 0) || (ms < max))) {
char ms_str[10];
ast_log(LOG_DEBUG, "Found qualified token of %d ms\n", ms);
/* Save detected talk time (in milliseconds) */
sprintf(ms_str, "%d", ms );
pbx_builtin_setvar_helper(chan, "TALK_DETECTED", ms_str);
if (ast_exists_extension(chan, chan->context, "talk", 1, chan->cid.cid_num)) {
strncpy(chan->exten, "talk", sizeof(chan->exten) -1 );
chan->priority = 0;

2
channels/chan_sip.c
View File

@@ -2638,7 +2638,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
p->novideo = 1;
while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') {
if ((sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1) ||
(sscanf(m, "audio %d/%d RTP/AVP %n", &x, &y, &len) == 1)) {
(sscanf(m, "audio %d/%d RTP/AVP %n", &x, &y, &len) == 2)) {
portno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
ast_rtp_pt_clear(p->rtp);