[VoiceOps] Preventing random SIP connections to handsets

Alex Balashov abalashov at evaristesys.com
Fri Nov 20 15:19:05 EST 2015

I was getting ghost ringing into my Polycom because my router sensibly 
remaps phone:5060 to WAN_IP:5060. My solution was to switch to SIP TCP.

On 11/20/2015 03:14 PM, Carlos Alvarez wrote:

> We're starting to see customers who get random arbitrary ringing caused
> by a random connection attempt from the internet.  Most of our customers
> have Cisco routers with full-cone NAT, so it's easy to do that.  We
> don't reinvite handsets, we proxy the media, so we've considered using
> restricted NAT instead.  If we can figure out how, we can't find any
> documentation on how to do it, and don't have a response to our Cisco
> TAC case on it yet.
> But I figured I'd ask if others have come up with better solutions.  I
> know there are a few authentication options in the phones themselves,
> but they seem to vary greatly by vendor and even by model.  I like to do
> things as simply and system-wide as possible.  We primarily sell
> Grandstream, and we support Cisco/Linksys SPA as well as Polycom IP
> series (not VVX).
> We're an Asterisk-based hosted service provider.
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Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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