Sure, so we have a number of users that dial into a room, say 4600, they receive audio and video from the moderator but don't send anything themselves because of mute|vmute. My customer wants to instead of calling in as a moderator, to instead send content to the room via SlingStudio like they do with Facebook and Youtube. Sling Studio sends a stream as rtmp://{my ip}/{room} I was hoping I could do that with mod_rtmp but was not able to make it work. So I am looking for another way to send video from the RTMP server to the FreeSWITCH room. I have found SIP to RTMP software, but I am trying to do the opposite, I need to get from RTMP to SIP room.
<extension name="Simple add-caller conf: member">
<condition field="destination_number" expression="^(46\d{2})$">
<action application="answer"/>
<action application="conference" data="$1@default+flags{mute|vmute}"/>
</condition>
</extension>
<extension name="Simple add-caller conf: moderator">
<condition field="destination_number" expression="^\*(46\d{2}$)">
<action application="answer"/>
<action application="digit_action_set_realm" data="moderator"/>
<action application="conference" data="$1@simple+flags{moderator}"/>
</condition>
</extension>
><>
nathan stratton