[Freeswitch-users] Mixing zap and sip users in the directory

While trying to uncover my issues with no sound on a zap channel, somewhere in the archive I came across the mention of mixing analog and digital extensions in the directory, is there anything special that needs to be done for this? Also, to answer an incoming call from an fxo channel, can I simply transfer it: <include> <extension name="Openzap_in"> <condition field="source" expression="mod_openzap"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="2000 XML default"/> </condition> </extension> </include> Or do I have to answer and do other things? Thanks! jlc