[App_rpt-users] SIP VoIP for Asterisk

[Dwaine Garden VE3GIF]

I changed the port.   They just kept scanning every IP port for the IP address.   I had to ask Rogers for another dhcp IP address.

The whole time they never got into the box.  LOL....

Steve Agee <n5zua at earthlink.net> wrote:

>I change my SIP port from the default 5060 to 15060 and have not had any 
>issues since. It seems that (most of the time) they only try the default.
>
>N5ZUA
>
>----- Original Message ----- 
>From: "Dwaine Garden VE3GIF" <DwaineGarden at rogers.com>
>To: "Bill South" <wbs099 at yahoo.com>
>Cc: <app_rpt-users at ohnosec.org>
>Sent: Thursday, September 05, 2013 11:13 AM
>Subject: Re: [App_rpt-users] SIP VoIP for Asterisk
>
>
>> 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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>