add manual_rtp_bug gen_one_gen_all to prevent rtp passthru to break the *S* word..

src/include/switch_types.h

@@ -685,7 +685,7 @@ typedef enum {
 	 */
 
 
-	RTP_BUG_ACCEPT_ANY_PACKETS = (1 << 7)
+	RTP_BUG_ACCEPT_ANY_PACKETS = (1 << 7),
 
 	/*
 	  Oracle's Contact Center Anywhere (CCA) likes to use a single RTP socket to send all its outbound audio.
@@ -694,6 +694,21 @@ typedef enum {
 	*/
 
 
+	RTP_BUG_GEN_ONE_GEN_ALL = (1 << 8)
+
+	/*
+	  Some RTP endpoints (and by some we mean *cough* _SONUS_!) do not like it when the timestamps jump forward or backwards in time.
+	  So say you are generating a file that says "please wait for me to complete your call, or generating ringback"
+	  Now you place and outbound call and you are bridging.  Well, while you were playing the file, you were generating your own RTP timestamps.
+	  But, now that you have a remote RTP stream, you'd rather send those timestamps as-is in case they will be fed to a remote jitter buffer......
+	  Ok, so this causes the audio to completely fade out despite the fact that we send the mark bit which should give them heads up its happening.
+
+	  Sigh, This flag will tell FreeSWITCH that if it ever generates even one RTP packet itself, to continue to generate all of them and ignore the
+	  actual timestamps in the frames.
+
+	 */
+
+
 } switch_rtp_bug_flag_t;
 
 #ifdef _MSC_VER

src/mod/applications/mod_dptools/mod_dptools.c

@@ -3730,8 +3730,8 @@ static int next_file(switch_file_handle_t *handle)
 	}
 
 	handle->samples = context->fh.samples;
-	handle->samplerate = context->fh.samplerate;
-	handle->channels = context->fh.channels;
+	//handle->samplerate = context->fh.samplerate;
+	//handle->channels = context->fh.channels;
 	handle->format = context->fh.format;
 	handle->sections = context->fh.sections;
 	handle->seekable = context->fh.seekable;

src/mod/endpoints/mod_sofia/sofia_glue.c

@@ -6577,6 +6577,14 @@ void sofia_glue_parse_rtp_bugs(switch_rtp_bug_flag_t *flag_pole, const char *str
 	if (switch_stristr("~ACCEPT_ANY_PACKETS", str)) {
 		*flag_pole &= ~RTP_BUG_ACCEPT_ANY_PACKETS;
 	}
+
+	if (switch_stristr("GEN_ONE_GEN_ALL", str)) {
+		*flag_pole |= RTP_BUG_GEN_ONE_GEN_ALL;
+	}
+
+	if (switch_stristr("~GEN_ONE_GEN_ALL", str)) {
+		*flag_pole &= ~RTP_BUG_GEN_ONE_GEN_ALL;
+	}
 }
 
 char *sofia_glue_gen_contact_str(sofia_profile_t *profile, sip_t const *sip, sofia_dispatch_event_t *de, sofia_nat_parse_t *np)

src/switch_rtp.c

@@ -4175,6 +4175,11 @@ SWITCH_DECLARE(int) switch_rtp_write_frame(switch_rtp_t *rtp_session, switch_fra
 
 	fwd = (switch_test_flag(rtp_session, SWITCH_RTP_FLAG_RAW_WRITE) && switch_test_flag(frame, SFF_RAW_RTP)) ? 1 : 0;
 
+	if (!fwd && switch_test_flag(rtp_session, SWITCH_RTP_FLAG_RAW_WRITE) && (rtp_session->rtp_bugs & RTP_BUG_GEN_ONE_GEN_ALL)) {
+		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Generating RTP locally but timestamp passthru is configured, disabling....\n");
+		switch_clear_flag(rtp_session, SWITCH_RTP_FLAG_RAW_WRITE);
+	}
+
 	switch_assert(frame != NULL);
 
 	if (switch_test_flag(frame, SFF_CNG)) {