How to secure access to PSTN line through Linksys gateway?

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How to secure access to PSTN line through Linksys gateway?

Postby vincentdelporte » Mon Nov 20, 2006 5:03 pm

Hello

I successfully hooked up a Linksys 3102 SIP gateway (http://www.voip-info.org/users/683/2168 ... 02_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.
vincentdelporte
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Postby RedStapler » Mon Nov 20, 2006 8:34 pm

Router: I suppose you are fine if you do not open any incoming ports, especially 5060 which is used by VoIP phones to connect to Asterisk.
Asterisk: Then there's incoming calls. This is independent of the network infrastructure. You need to separate incoming/outgoing/internal contexts in your dialplan. Includes are used for that.

Code: Select all
[incoming]
exten => s,1,Answer()
...
[internal]
include => outbound
...

[outbound]
exten => _9NXXXXXX,1,Dial(...


The book "Asterisk, the future of telephony" explains this.
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Postby lee_is_me » Mon Nov 20, 2006 9:00 pm

And a great book too. The perfect reference tool for me.
Lee

"I figured the best way to learn Asterisk is to write a cool Windows based GUI for it. It's now in beta and I think I have learned 1% of what there is to learn."
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Postby vincentdelporte » Tue Nov 21, 2006 7:40 am

RedStapler wrote:Router: I suppose you are fine if you do not open any incoming ports, especially 5060 which is used by VoIP phones to connect to Asterisk.


Unfortunately, I have to open UDP 5060 as remote users must register with the Asterisk server to be part of a group ring that is called when PSTN calls come in through the Linksys.

RedStapler wrote:Asterisk: Then there's incoming calls. This is independent of the network infrastructure. You need to separate incoming/outgoing/internal contexts in your dialplan. Includes are used for that.


OK. I'll add contexts to that purpose.

Thank you.
vincentdelporte
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Posts: 30
Joined: Thu Dec 21, 2006 10:56 pm

Postby RedStapler » Tue Nov 21, 2006 8:02 am

vincentdelporte wrote:remote users must register with the Asterisk server to be part of a group ring that is called when PSTN calls come in through the Linksys.
I see. Hmm, I did not set up VPN for VoIP ever, but here's an article which might give you a start:
http://www.voip-news-net.com/2006/02/se ... our_d.html
Hope this helps.
RedStapler
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Postby vincentdelporte » Wed Nov 22, 2006 7:08 am

RedStapler wrote:I see. Hmm, I did not set up VPN for VoIP ever, but here's an article which might give you a start


Thanks. I don't know if it's worth it to go all the way and deploy VPN's just to route SIP calls, but I'll check it out.
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