[VoiceOps] Preventing random SIP connections to handsets
robert.j at bendtel.com
Fri Nov 20 15:35:23 EST 2015
On 11/20/2015 12:14 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.
> VoiceOps mailing list
> VoiceOps at voiceops.org
This may be dependent upon the Cisco router in question, but when we
deploy routers we always set the ACL to only allow SIP communications
from our SBC. - When customers provide their own, we recommend the same
Central Oregon's Own Telephone and Internet Service Provider
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