Hi William, Turn on sip debugging on your Asterisk server. Maybe the 404 is what Asterisk is sending back to your Freeswitch. You probably need to configure a SIP channel in Asterisk to allow for calls from the freeswitch to come in on. eg [freeswitch] type=peer context=freeswitch-in insecure=very host=<freeswitchIP> Regards, On 8/30/06, William Piper <william.piper@gmail.com> wrote:
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,
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