Good Afternoon, I currently have a video conference which users connect to via the Verto library. I was looking to enhance this conference by also allowing users to join audio-wise to the video conference by making a bridge/call to my asterisk server which hosts audio conference rooms. Currently I have the conference_set_auto_outcall making the call via sip to the asterisk server, but ran into a couple of issues after asterisk and the freeswitch servers establish the call. The first issue, though it may be on purpose, is that when I use the ghost flag for the member it seems RTP is never delivered to my verto endpoint from the sip call to the asterisk as well as the other way around. Though I clearly see freeswitch is recognizing energy detection from my asterisk conference, as I see the talking flag appear but no audio is heard from it on the verto client. It is the same the other way around, where I will talk via a mic on my verto client, but never hear it on the asterisk side of things. Though for asterisk it does not even seem to be sending audio to the asterisk server. I'd prefer to use the ghost flag as I do with screen sharing so that the audio bridge does not count as a user for the video conference, of course I can always get around this by just bumping up the max users by one too if the user decides to use the bridge. The second issue might be more of a misconfiguration issue, but I am a bit stuck on where to check. For other times when I don't have the ghost or any flag set for the outbound bridge call, things work as expected. I can transmit and hear audio from both ends to the other one without any issue. Though even though Freeswitch knows the bridged call has no video since has_video is set to false in conference <Room> xml_list, it seems to still try to use that as the presenter / floor occasionally. During these times I get the following in fs_cli constantly 2017-05-09 17:10:05.742387 [WARNING] switch_core_media.c:11459 sofia/internal/352650@asterisk-server has no video codec wher asterisk server is actually the IP/dns host name of the asterisk server I develop with. Ideally if I were not to use the ghost flag I'd want it to treat this bridged call as audio only and not constantly spew that warning message. Some ideas I haven't tested yet as I am not too familiar with freeswitch besides verto/mod_conference a bit, was to see if I could have it join the conference with an audio only profile maybe, or somehow set the SIP not to try offering video maybe. Any ideas? Here is the profile and dialplan I was using to test Conference profile (returned from mod_xml_curl): <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> Thanks in advance, Zach