[App_rpt-users] SIP VoIP for Asterisk

[Steve Agee]

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
>>
>>
>>_______________________________________________
>>App_rpt-users mailing list
>>App_rpt-users at ohnosec.org
>>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users