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How to secure access to PSTN line through Linksys gateway?

 
 
Vincent Delporte
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      11-20-2006
Hello

I successfully hooked up a Linksys 3102 SIP gateway
(http://www.voip-info.org/users/683/2...PA3102_lrg.jpg)
to an Asterisk server, but since it's connected to a PSTN line, I must
make sure it cannot be used by unauthorized users from the Net.
Actually, even legit users with an account on the Asterisk server
shouldn't be able to use it (outgoing calls should go through a VoIP
provider instead).

Does someone know what steps must be taken in Asterisk and on the
Linksys to secure access to the Linksys box?

Thank you.
 
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Jonathan Roberts
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      11-21-2006
Vincent Delporte wrote:
> Hello
>
> I successfully hooked up a Linksys 3102 SIP gateway
> (http://www.voip-info.org/users/683/2...PA3102_lrg.jpg)
> to an Asterisk server, but since it's connected to a PSTN line, I must
> make sure it cannot be used by unauthorized users from the Net.
> Actually, even legit users with an account on the Asterisk server
> shouldn't be able to use it (outgoing calls should go through a VoIP
> provider instead).
>
> Does someone know what steps must be taken in Asterisk and on the
> Linksys to secure access to the Linksys box?
>
> Thank you.


Have you set it up as its own trunk in Asterisk?

Jonathan

PS-- is this 'plain vanilla' Asterisk or TrixBox/freepbx?
 
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Enzo Michelangeli
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      11-21-2006
"Vincent Delporte" <(E-Mail Removed)> wrote in message
news:(E-Mail Removed)...
> Hello
>
> I successfully hooked up a Linksys 3102 SIP gateway
> (http://www.voip-info.org/users/683/2...PA3102_lrg.jpg)
> to an Asterisk server, but since it's connected to a PSTN line, I must
> make sure it cannot be used by unauthorized users from the Net.
> Actually, even legit users with an account on the Asterisk server
> shouldn't be able to use it (outgoing calls should go through a VoIP
> provider instead).
>
> Does someone know what steps must be taken in Asterisk and on the
> Linksys to secure access to the Linksys box?


It's very easy to prevent outgoing calls initiated by Asterisk: just don't
put anywhere in the dialplan a "Dial()" statement referencing the 3102 (or
if you do, don't make the context where it occurs accessible by regular
users).

If you worry about direct SIP sessions from SIP clients to the 3101, you
might register the 3102 to an Asterisk account and set "Make Call Without
Reg:" to "No" in the "PSTN Line" screen.

Or, if nobody should be authorized to dial out, just disable the
"VoIP-to-PSTN" function. In the SPA-3100 you can set the "VoIP-To-PSTN
Gateway Enable:" to "No" in the same "PSTN Line" screen: the 3102 should be
similar.

Enzo


 
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Vincent Delporte
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      11-21-2006
On Mon, 20 Nov 2006 18:25:08 -0600, Jonathan Roberts
<(E-Mail Removed)> wrote:
>Have you set it up as its own trunk in Asterisk?


No idea I just added an account for this device in sip.conf.

>PS-- is this 'plain vanilla' Asterisk or TrixBox/freepbx?


plain vanilla (with hot fudge on top, I confess).
 
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Jonathan Roberts
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      11-21-2006
Vincent Delporte wrote:
> On Mon, 20 Nov 2006 18:25:08 -0600, Jonathan Roberts
> <(E-Mail Removed)> wrote:
>> Have you set it up as its own trunk in Asterisk?

>
> No idea I just added an account for this device in sip.conf.
>
>> PS-- is this 'plain vanilla' Asterisk or TrixBox/freepbx?

>
> plain vanilla (with hot fudge on top, I confess).


Looks like Enzo got to you first Good luck!

Jonathan
 
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Vincent Delporte
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      11-21-2006
On Tue, 21 Nov 2006 08:26:03 +0800, "Enzo Michelangeli"
<(E-Mail Removed)-ip.com> wrote:
(snip)

Thanks a lot I'll check those three points.
 
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