Shortly after posting my original message it suddenly dawned on me that my soft-client might be attempting to dial into my server using port 5060 which i recalled was used, by default, for users in the default context, not the public. Tacking :5080 onto the end of the SIP URI i was dialing gave me a whole different set of errors. After reading through the log messages and trying out some different configuration i managed to get the basics working (accepting a call from an external domain and passing it through to an internal user or directing to voicemail if user isn't online). I'll post my config here incase anyone else ever has this particular problem. conf/directory/default/bert.xml: <include> <user id="bert" mailbox="2378"> <params> <param name="password" value="password"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="accountcode" value="2378"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Bert"/> <variable name="effective_caller_id_number" value="2378"/> </variables> </user> </include> Added to conf/dialplan/public.xml: <extension name="bert"> <condition field="destination_number" expression="^(bert|2378)$"> <action application="transfer" data="bert XML default"/> </condition> </extension> Create the file conf/dialplan/extensions/bert.xml: <include> <extension name="bert"> <condition field="destination_number" expression="^(bert|2378)$"> <action application="set" data="dialed_ext=$1"/> <action application="export" data="dialed_ext=$1"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default $${domain} ${dialed_ext}"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <anti-action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <anti-action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <anti-action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <anti-action application="set" data="transfer_ringback=${uk-ring}"/> <anti-action application="set" data="call_timeout=30"/> <!-- <anti-action application="set" data="sip_exclude_contact=${network_addr}"/> --> <anti-action application="set" data="hangup_after_bridge=true"/> <!--<anti-action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/> <anti-action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/> <anti-action application="bridge" data="user/${dialed_ext}@$${domain}"/> <anti-action application="answer"/> <!--<anti-action application="send_display" data="Voicemail for ${dialed_ext}"/>--> <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension> </include> This seems to work well, although the IVR before leaving a vm seems a bit choppy (perhaps due to being on a Xen VPS?). The only question i still have regarding this bit is whether it would be better to switch the ports around so that FreeSWITCH accepts calls from external domains on port 5060 and registrations within the default context on port 5080 or do the majority of SIP servers look up and adhere to what's specified in a domain's SRV records?