[Freeswitch-users] Initial Startup

[Vikram Rangnekar]

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 at 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 at 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 at 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 at 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 at 70.159.49.41
>
> If it's not obvious enough, I'm a super newbie.
>
> Thanks,
>
> bp
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-- 
Vikram Rangnekar