So I tested with Freeswitch 1.6.17 as Brian suggested earlier, and it seems that the errors and warnings I saw still occur. To summarize the original post I made, I have my Freeswitch server which supplies the server for video conferences using Verto. I have it bridging to an Asterisk server (testing with Asterisk 13) to allow PSTN calls to be bridged to Freeswitch via an Asterisk MeetMe conference room. Freeswitch gets its dialplan for Verto via mod_xml_curl and sends the call to Asterisk via the conference auto outcall application. When I set the flags for this bridged call to ghost it seems Freeswitch is receiving the RTP, as I see verto say "Talking" properly, but I never hear anything from my verto client connected to the video conference room. This does not happen when the ghost flag is removed, as I can speak from either end and hear it on the other one. Additionally when the bridged call initially joins the conference, it seems to be being placed as the presenter even though I have the flags so that without video it can't be on the canvas. Such I get the following warning flooding the console: 2017-05-10 12:51:57.505225 [WARNING] switch_core_media.c:11774 sofia/internal/352650@asterskdev has no video codec This warning stops both when the floor is not trying to be set to the audio bridge, or when the floor is locked to another presenter. It recognizes properly you can't set the vid floor to that bridged call in fs_cli as it says in console: 2017-05-10 13:33:18.125230 [DEBUG] conference_event.c:110 conf 3525236 CMD mod_verto [vid-floor] 0002 2017-05-10 13:33:18.125230 [ERR] conference_api.c:1770 Channel sofia/internal/352650@hasterisk-dev does not have video capability! 2017-05-10 13:33:18.125230 [ALERT] conference_event.c:316 RES [3525236 vid-floor 0002 force][] I've included both the profile and dialplan returned by my web application for mod_xml_curl. Dialplan: <document type="freeswitch/xml"> <section name="configuration"> <configuration name="conference.conf" description="Conf Profiles"> <profiles> <profile name="3525236"> <param name="domain" value="$${domain}"/> <param name="rate" value="48000"/> <param name="channels" value="2"/> <param name="interval" value="20"/> <param name="energy-level" value="200"/> <param name="muted-sound" value="conference/conf-muted.wav"/> <param name="unmuted-sound" value="conference/conf-unmuted.wav"/> <param name="alone-sound" value="conference/conf-alone.wav"/> <param name="hold-music" value="$${hold_music}"/> <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/> <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/> <param name="kicked-sound" value="conference/conf-kicked.wav"/> <param name="locked-sound" value="conference/conf-locked.wav"/> <param name="is-locked-sound" value="conference/conf-is-locked.wav"/> <param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/> <param name="pin-sound" value="conference/conf-pin.wav"/> <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/> <param name="caller-id-name" value="$${outbound_caller_name}"/> <param name="caller-id-number" value="$${outbound_caller_id}"/> <param name="comfort-noise" value="false"/> <param name="conference-flags" value="video-floor-only|rfc-4579|livearray-sync|minimize-video-encoding|livearray-json-status|video-required-for-canvas|video-mute-exit-canvas"/> <param name="video-mode" value="passthrough"/> <param name="video-layout-name" value="1x1"/> <param name="video-layout-name" value="group:grid"/> <param name="video-canvas-size" value="1920x1080"/> <param name="video-canvas-bgcolor" value="#333333"/> <param name="video-layout-bgcolor" value="#000000"/> <param name="video-codec-bandwidth" value="1mb"/> <param name="video-fps" value="15"/> <param name="max-members-sound" value="conference/conf-conference_is_full.wav"/> <param name="endconf-grace-time" value="30"/> <param name="max-members" value="30"/> </profile> </profiles> </configuration> </section> </document> Dialplan(Returned from mod_xml_curl): <document type="freeswitch/xml"> <section name="dialplan" description="Dialplan routing"> <context name="default"> <extension name="3525236"> <condition field="destination_number" expression="^3525236-moderator$"> <action application="set" data="api_hangup_hook=system ROOM=3525236 VCONFID=239 /usr/local/bin/flip_conference_flag.sh"/> <action application="answer"/> <action application="set" data="conference_auto_outcall_timeout=5"/> <action application="set" data="conference_utils_auto_outcall_flags=ghost"/> <action application="set" data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/> <action application="set" data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/> <action application="set" data="conference_auto_outcall_profile=3525236"/> <action application="conference_set_auto_outcall" data="sofia/internal/352650@asterisk_server_hostname"/> <action application="sleep" data="2000"/> <action application="conference" data="3525236@3525236+flags{moderator|endconf}"/> <action application="hangup"/> </condition> </extension> </context> </section> </document>