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@rogers.com> To: "Bill South" <wbs099@yahoo.com> Cc: <app_rpt-users@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@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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