List, I tried asking this on IRC but email seemed easier to explain everything: I'm trying to setup Freeswitch to pass traffic strictly from SIP to SIP. I want to allow traffic only from static IP's that I define and block all others. When a call comes in from an IP that is allowed, the call should process and be sent out to one of our Asterisk servers that has PSTN connectivity. I guess my question is... How do I do this? Here is what I have so far. From what I understand, it should pretty much be saying: IF IP == 70.159.49.36 AND dst == 13523985807 then call 3523985807@66.118.164.51. <extension name="skynet" continue="true"> <condition field="network_addr" expression="70.159.49.36"/> <condition field="destination_number" expression="13523985807"> <action application="bridge" data=" exosip/3523985807@66.118.164.51"/> </condition> </extension> Am I even half way correct in my XML? I get 404 not found as shown below with this setup. 0.000000 70.159.49.36 -> 70.159.49.41 SIP/SDP Request: INVITE sip:13523985807@70.159.49.41, with session description 0.001979 70.159.49.41 -> 70.159.49.36 SIP Status: 100 Trying 10.001208 70.159.49.41 -> 70.159.49.36 SIP Status: 404 Not Found 10.001783 70.159.49.36 -> 70.159.49.41 SIP Request: ACK sip:13523985807@70.159.49.41 If it's not obvious enough, I'm a super newbie. Thanks, bp