[Freeswitch-users] Initial Startup

[William Piper]

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 at 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 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
>
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