disconnect callbacks from rtp session when AudioStream is destroyed, otherwise...

disconnect callbacks from rtp session when AudioStream is destroyed, otherwise they remain in the rtp session, which is kept for further updates of the session (reINVITE)

src/voip/audiostream.c

@@ -980,6 +980,8 @@ void audio_stream_stop(AudioStream * stream){
 			}
 		}
 	}
+	rtp_session_signal_disconnect_by_callback(stream->ms.sessions.rtp_session,"telephone-event",(RtpCallback)on_dtmf_received);
+	rtp_session_signal_disconnect_by_callback(stream->ms.sessions.rtp_session,"payload_type_changed",(RtpCallback)mediastream_payload_type_changed);
 	audio_stream_free(stream);
 	ms_filter_log_statistics();
 }
@@ -1025,6 +1027,8 @@ void audio_stream_enable_zrtp(AudioStream *stream, OrtpZrtpParams *params){
 	else if (!stream->ms.sessions.is_secured)
 		ortp_zrtp_reset_transmition_timer(stream->ms.sessions.zrtp_context,stream->ms.sessions.rtp_session);
 }
+
 bool_t audio_stream_zrtp_enabled(const AudioStream *stream) {
 	return stream->ms.sessions.zrtp_context!=NULL;
 }
+