Thanks, Yea, I have that setup. I've been working with Asterisk for a while... I'm sure I have it setup right. The ethereal trace on the previous email shows the 404 coming from the .41 address (.36 is the asterisk server sending the call). That is the Freeswitch address... so freeswitch is unable to find the route. Just for fun, I tried calling some other ones that are in the dialplan and they fail too. I tried 888, which should direct me to the FS conference room, but it also gave me the 404 error. For FS to give the 404 error, it seems to tell me that it is working... it just can't find the destination. I have IPtables turned off for testing but no difference. Any other thoughts? Thanks, bp On 8/30/06, Vikram Rangnekar <vr@udel.edu> wrote:
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,
bp _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Vikram Rangnekar
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