diff --git a/apps/app_disa.c b/apps/app_disa.c
index 576398b4fb..d93bb1485b 100644
--- a/apps/app_disa.c
+++ b/apps/app_disa.c
@@ -160,8 +160,12 @@ static int disa_exec(struct ast_channel *chan, void *data)
 			/* answer */
 			ast_answer(chan);
 		}
-	} else
+	} else {
 		special_noanswer = 1;
+		if (chan->_state != AST_STATE_UP) {
+			ast_indicate(chan, AST_CONTROL_PROGRESS);
+		}
+	}
 
 	ast_debug(1, "Context: %s\n",args.context);
 
diff --git a/apps/app_playback.c b/apps/app_playback.c
index ed40194c97..0be2c2e48f 100644
--- a/apps/app_playback.c
+++ b/apps/app_playback.c
@@ -426,9 +426,13 @@ static int playback_exec(struct ast_channel *chan, void *data)
 		if (option_skip) {
 			/* At the user's option, skip if the line is not up */
 			goto done;
-		} else if (!option_noanswer)
+		} else if (!option_noanswer) {
 			/* Otherwise answer unless we're supposed to send this while on-hook */
 			res = ast_answer(chan);
+		} else {
+			ast_indicate(chan, AST_CONTROL_PROGRESS);
+		}
+
 	}
 	if (!res) {
 		char *back = args.filenames;
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 9d81ead2f1..e2848bbd17 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -762,6 +762,8 @@ static int pedanticsipchecking;		/*!< Extra checking ?  Default off */
 static int autocreatepeer;		/*!< Auto creation of peers at registration? Default off. */
 static int global_match_auth_username;		/*!< Match auth username if available instead of From: Default off. */
 static int global_relaxdtmf;		/*!< Relax DTMF */
+static int global_prematuremediafilter;	/*!< Enable/disable premature frames in a call (causing 183 early media) */
+static int global_relaxdtmf;			/*!< Relax DTMF */
 static int global_rtptimeout;		/*!< Time out call if no RTP */
 static int global_rtpholdtimeout;	/*!< Time out call if no RTP during hold */
 static int global_rtpkeepalive;		/*!< Send RTP keepalives */
@@ -5569,9 +5571,11 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
 				    !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
 				    !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
 					ast_rtp_new_source(p->rtp);
-					p->invitestate = INV_EARLY_MEDIA;
-					transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
-					ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
+					if (!global_prematuremediafilter) {
+						p->invitestate = INV_EARLY_MEDIA;
+						transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
+						ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
+					}
 				} else if (p->t38.state == T38_ENABLED) {
 					change_t38_state(p, T38_DISABLED);
 					transmit_reinvite_with_sdp(p, FALSE, FALSE);
@@ -14751,6 +14755,7 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
  	ast_cli(a->fd, "  Timer T1:               %d\n", global_t1);
 	ast_cli(a->fd, "  Timer T1 minimum:       %d\n", global_t1min);
  	ast_cli(a->fd, "  Timer B:                %d\n", global_timer_b);
+	ast_cli(a->fd, "  No premature media:     %s\n", global_prematuremediafilter ? "Yes" : "No");
 
 	ast_cli(a->fd, "\nDefault Settings:\n");
 	ast_cli(a->fd, "-----------------\n");
@@ -22716,6 +22721,7 @@ static int reload_config(enum channelreloadreason reason)
 	snprintf(global_useragent, sizeof(global_useragent), "%s %s", DEFAULT_USERAGENT, ast_get_version());
 	snprintf(global_sdpsession, sizeof(global_sdpsession), "%s %s", DEFAULT_SDPSESSION, ast_get_version());
 	snprintf(global_sdpowner, sizeof(global_sdpowner), "%s", DEFAULT_SDPOWNER);
+	global_prematuremediafilter = TRUE;
 	ast_copy_string(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime));
 	ast_copy_string(global_realm, S_OR(ast_config_AST_SYSTEM_NAME, DEFAULT_REALM), sizeof(global_realm));
 	ast_copy_string(default_callerid, DEFAULT_CALLERID, sizeof(default_callerid));
@@ -22877,6 +22883,8 @@ static int reload_config(enum channelreloadreason reason)
 			ast_set2_flag(&global_flags[1], i || ast_true(v->value), SIP_PAGE2_RTAUTOCLEAR);
 		} else if (!strcasecmp(v->name, "usereqphone")) {
 			ast_set2_flag(&global_flags[0], ast_true(v->value), SIP_USEREQPHONE);	
+		} else if (!strcasecmp(v->name, "prematuremedia")) {
+			global_prematuremediafilter = ast_true(v->value);
 		} else if (!strcasecmp(v->name, "relaxdtmf")) {
 			global_relaxdtmf = ast_true(v->value);
 		} else if (!strcasecmp(v->name, "vmexten")) {
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 7bfcda9875..2b260eb257 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -208,6 +208,13 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;relaxdtmf=yes                  ; Relax dtmf handling
 ;trustrpid = no                 ; If Remote-Party-ID should be trusted
 ;sendrpid = yes                 ; If Remote-Party-ID should be sent
+;prematuremedia=no		; Some ISDN links send empty media frames before 
+				; the call is in ringing or progress state. The SIP 
+				; channel will then send 183 indicating early media
+				; which will be empty - thus users get no ring signal.
+				; Setting this to "no" will stop any media before we have
+				; call progress. Default is "yes".
+
 ;progressinband=never           ; If we should generate in-band ringing always
                                 ; use 'never' to never use in-band signalling, even in cases
                                 ; where some buggy devices might not render it
diff --git a/main/pbx.c b/main/pbx.c
index a072c9a415..5fa6cf6ba4 100644
--- a/main/pbx.c
+++ b/main/pbx.c
@@ -8238,6 +8238,8 @@ static int pbx_builtin_background(struct ast_channel *chan, void *data)
 		} else if (!ast_test_flag(&flags, BACKGROUND_NOANSWER)) {
 			res = ast_answer(chan);
 		}
+		/* Send progress control frame to start early media */
+		ast_indicate(chan, AST_CONTROL_PROGRESS);
 	}
 
 	if (!res) {