I am trying to get media stats on my call, using the following dialplan: <extension name="announcement"> <condition field="${sip_req_user}" expression="^annc$"/> <condition field="${sip_req_params}" expression="^play=(.*)$"> <action application="answer"/> <action application="set" data="api_result=${uuid_set_media_stats(${uuid})}"/> <action application="sleep" data="1000"/> <action application="playback" data="http_cache://http://wcs-icr.s3.amazonaws.com/$1"/> <action application="set" data="api_result=${uuid_dump(${uuid})"/> <action application="log" data="incoming bytes ${rtp_audio_in_raw_bytes}"/> <action application="log" data="incoming bytes ${rtp_audio_out_raw_bytes}"/> <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/> </condition> </extension> However, the data that is dumped shows packet counts of zero. The prompt is playing, so we are certainly sending packets out. Any ideas? Here is some of the output from the uuid_dump: variable_endpoint_disposition: ANSWER variable_rtp_audio_in_raw_bytes: 0 variable_rtp_audio_in_media_bytes: 0 variable_rtp_audio_in_packet_count: 0 variable_rtp_audio_in_media_packet_count: 0 variable_rtp_audio_in_skip_packet_count: 0 variable_rtp_audio_in_jitter_packet_count: 0 variable_rtp_audio_in_dtmf_packet_count: 0 variable_rtp_audio_in_cng_packet_count: 0 variable_rtp_audio_in_flush_packet_count: 0 variable_rtp_audio_in_largest_jb_size: 0 variable_rtp_audio_in_jitter_min_variance: 0.00 variable_rtp_audio_in_jitter_max_variance: 0.00 variable_rtp_audio_in_jitter_loss_rate: 0.00 variable_rtp_audio_in_jitter_burst_rate: 0.00 variable_rtp_audio_in_mean_interval: 0.00 variable_rtp_audio_in_flaw_total: 0 variable_rtp_audio_in_quality_percentage: 100.00 variable_rtp_audio_in_mos: 4.50 variable_rtp_audio_out_raw_bytes: 0 variable_rtp_audio_out_media_bytes: 0 variable_rtp_audio_out_packet_count: 0 variable_rtp_audio_out_media_packet_count: 0 variable_rtp_audio_out_skip_packet_count: 0 variable_rtp_audio_out_dtmf_packet_count: 0 variable_rtp_audio_out_cng_packet_count: 0 variable_rtp_audio_rtcp_packet_count: 0 variable_rtp_audio_rtcp_octet_count: 0 EXECUTE sofia/drachtio_mrf/5083084809@68.64.80.16 log(incoming bytes 0) 2016-06-17 10:59:28.757502 [DEBUG] mod_dptools.c:1713 bytes 0 EXECUTE sofia/drachtio_mrf/5083084809@68.64.80.16 log(incoming bytes 0) 2016-06-17 10:59:28.757502 [DEBUG] mod_dptools.c:1713 bytes 0