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
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>
>
>


--
Vikram Rangnekar

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