Merged revisions 162204 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7 lines
  
  Make sure that the timestamp for DTMF is not the same as the previous voice frame and do not send audio when transmitting DTMF as this confuses some equipment.
  (closes issue #13209)
  Reported by: ip-rob
  Patches:
        13209.diff uploaded by file (license 11)
  Tested by: ip-rob, bujones
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162205 65c4cc65-6c06-0410-ace0-fbb531ad65f3

main/rtp.c

@@ -3050,6 +3050,7 @@ int ast_rtp_senddigit_begin(struct ast_rtp *rtp, char digit)
 
 	rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
 	rtp->send_duration = 160;
+	rtp->lastdigitts = rtp->lastts + rtp->send_duration;
 	
 	/* Get a pointer to the header */
 	rtpheader = (unsigned int *)data;
@@ -3169,10 +3170,9 @@ int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit)
 				    ast_inet_ntoa(rtp->them.sin_addr),
 				    ntohs(rtp->them.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
 	}
+	rtp->lastts += rtp->send_duration;
 	rtp->sending_digit = 0;
 	rtp->send_digit = 0;
-	/* Increment lastdigitts */
-	rtp->lastdigitts += 960;
 	rtp->seqno++;
 
 	return res;
@@ -3498,6 +3498,10 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
 	int pred;
 	int mark = 0;
 
+	if (rtp->sending_digit) {
+		return 0;
+	}
+
 	ms = calc_txstamp(rtp, &f->delivery);
 	/* Default prediction */
 	if (f->frametype == AST_FRAME_VOICE) {