Add TALK_DETECTED variable (bug #2628)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4139 65c4cc65-6c06-0410-ace0-fbb531ad65f3

apps/app_talkdetect.c

@@ -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;

channels/chan_sip.c

@@ -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);