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