For something like that you would need a dual-mode cellphone (something that does GSM/CDMA,etc on one side, and 802.11g w/ SIP on the other side). Then, when you're in your house, your phone would join the wifi network and register to a SIP controller (ala Asterisk). Then, when your landline rings, your cellphone would ring as well. The key is you need a handset that supports both GSM/CDMA, as well as 802.11g & SIP.

Have a look at this.

Get one of these:

http://www.myxlink.com/index.aspx

And keep the legacy landline handsets in the house. This way, no matter where you are in the house, whether or not the cellphone is with you, you can still make/receive calls - leveraging your cell minutes.

You can probably integrate that with an Asterisk VoIP system and get additional things like intercom, room-to-room dialing, etc.

My friends and I have a running game where we see how long we can keep them on the phone line. Play along, come up with funny objections, etc. The goal is to see how long you can keep them on the phone. I've only been able to get 20 minutes down - my wife's record is 26. Think of it as a honeypot - let them waste their time on opportunities that don't really exist.

I had similar problems. The best and easiest way to do this would be to rate-limit all TCP transmission/reception to some percentage less than your overall Internet bandwidth, say 80%. That way, all applications (FTP, BitTorrent, HTTP, etc.) won't take up all of your bandwidth, and thus kill your VoIP. The problem that you're having is that you're saturating your uplink or downlink, and that's what's delaying the VoIP packets. You only need to rate-limit TCP, since Vonage (and most VoIP out there) is UDP-based. You want to let the UDP through without touching it. By limiting TCP, you're effectively reserving UDP bandwidth for your VoIP.

You can do this with a fairly inexpensive Cisco router (buy an 800-series off ebay). You can probably do it with DD-WRT as well.

For the true gearheads out there, I worked out something similar, except that my system actually senses when a call is in process and makes appropriate TCP rate-limiting for the duration of the call. Once the call is complete, the full bandwidth becomes available for other things. It's a clever use of Snort, pfSense, and a Cisco router. I call it AATQoS or Application Aware Triggered Quality of Service.

Whatever you end up doing, just understand that your call quality is suffering due to traffic congestion, and if you can alleviate that congestion enough to let the VoIP UDP packets flow, then your call quality will sound great.