diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 4586ca67f6c7cef930612a571b614c8cb563e665..8841f6d6de935829aeaec8f527ed573a99cf8990 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -5732,16 +5732,6 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 	switch (ast_format_get_type(rtp->f.subclass.format)) {
 	case AST_MEDIA_TYPE_AUDIO:
 		rtp->f.frametype = AST_FRAME_VOICE;
-
-		/* The marker bit set on the voice packet indicates the start
-		 * of a new stream and a new time stamp. Need to reset the DTMF
-		 * last sequence number and the timestamp of the last END packet.
-		 */
-		if (mark) {
-			rtp->last_seqno = 0;
-			rtp->last_end_timestamp = 0;
-		}
-
 		break;
 	case AST_MEDIA_TYPE_VIDEO:
 		rtp->f.frametype = AST_FRAME_VIDEO;