[App_rpt-users] SIP VoIP for Asterisk

[Dwaine Garden VE3GIF]

The problem is you are not allowed by law to have a phone without unrestricted access to 911.  I had Metro Toronto police at my door explaining that even if I block 911 to any outside connections I would be breaking the law.  If you have a server on the internet with sip.  They have to able to connect to be able to call 911. 

I told the police it was retard.  They told me that was fine they will charge me.

Police told me that even if someone breaks into your house.  If there is a phone install,  the criminals better have access to dial 911 unrestrictive.

The hackers did not get into the box.  They were trying for months.   Got pissed off and changed their script to dial 911 constantly.  SIP and DID see a 911 call.  It dials it.  No questions asked.  No login or nothing.

The Police told me it was a huge problem.  SIP or DID are setup like a public pay phone.  Full access to 911.

It was an eye opener for me.   You learn something new everyday.  If I see someone asking about SIP or DID.  I let them know about my experience.

David KE6UPI <dshaw at ke6upi.com> wrote:

>I'm sorry Dwaine what are you talking about? Sorry If I don't understand what your talking about.
>
>I have both a public Asterisk server and a local Asterisk server. I have never had anyone connect and make a call that was not authenticated user.. Yes they try and fail2ban will block them. There are many way to stop unwanted hackers on your server. 
>
>
>As for dialing 911 just make a dial plain to route to space if you want. 
>
>Google "Asterisk Security"
>
>
>http://www.voip-info.org/wiki/view/Asterisk+security
>
>
>David
>
>
>
>
>On Thu, Sep 5, 2013 at 9:13 AM, Dwaine Garden VE3GIF <DwaineGarden at rogers.com> wrote:
>
>It works great until the hacks find the machine.  They port scan non stop.  Its especially fun when their scripts dial 911 constantly.  There is no way to turn off dialing 911 for SIP.
>
>
>Bill South <wbs099 at yahoo.com> wrote:
>
>>     I'm thinking of adding some type of SIP trunking or other VoIP service provider to my ACID Asterisk system to support in/out bound calling.  I've read some emails on the app_rpt reflector with names of providers mentioned, but I am looking for recommendations, as there are zillions of VoIP providers out there.  This is going to be used very sparingly so least-cost is a good thing, but good reliability and no bombardment with email adds by the provider is desired too.  I can easily get by with a single number, but may want to add additional DIDs later.  Thoughts?
>>
>>Bill
>>
>>
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