[App_rpt-users] SIP VoIP for Asterisk

[David KE6UPI]

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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