I wish to connect FS to * for interoperability testing and can't seem to get my configs correct. Has anyone done this already and if so can you post your configs? I can call from FS to the asterisk side via a dialplan entry where ^9(.*)$ passes to a gateway I've setup: <extension name="out9_asterisk"> <condition field="destination_number" expression="^9(.*)$"> <action application="bridge" data="sofia/gateway/asterisk/$1"/> </condition> </extension> sip_profiles/default/asterisk.xml : <include> <gateway name="asterisk"> <param name="username" value="freeswitch"/> <param name="realm" value="x.x.x.x"/> <param name="password" value="password"/> <param name="register" value="false"/> </gateway> </include> The problem I get with this is no audio either way. I am on rev 8099. TIA, Brian