Michael’s VoIP

I run VoIP at home exclusively. I currently have a headless Debian 3.1 Linux machine as my voice switch using the popular PBX software Asterisk. It’s subscribed to Free World Dial up, Stanaphone, Voicepulse and Packet8. I currently have 3 analog phones connected to the switch via a Cisco ATA-186 which provides 2 lines, and a firmware-hacked DTA-310 (packet8 device) which provides 1 line. These devices are all on a private network utilizing the 172.16.0.0 IP range and communicate only to the Asterisk PBX, which has 2 NICs on-board. The other NIC in the Asterisk machine has a NAT IP in the 192.168.x.x range assigned to it. A linksys wrt54g router connected to a standard roadrunner connection forwards ports 5060/UDP (SIP), 4569/UDP (IAX2), and 12000-12100/UDP (RTP) to it. I use an analog x101p zaptel card to connect to my packet8 device. Luckily, packet8 runs SIP traffic on port 5082, so there is not contention for port between the Asterisk server and the packet8 device. The linksys is using the sveasoft 3rd-party firmware which enables it to do a lot more than the stock linksys wrt54g firmware and most importantly packet shaping. This allows the router to prioritize VoIP traffic. I can’t have bandwidth-intensive applications like bittorrent wipe out my calls. With QoS, a voice call and a bandwidth-heavy application can co-exist nicely.

Asterisk PBX VoIP / POTS switch
Photo of the VoIP PBX

This setup gives me the ability to call from room to room, and make as many incoming and outgoing calls at the same time as I’d ever want. When someone calls from any of the subscribed services, all the phones that are not in use ring. Priority is given to packet8 ($24/mo flat rate) for outbound calls, then spillover to voicepulse (2.4c/min).

The next project I have is wiring the house properly and replacing some of the analog phones and FXS devices with self-contained SIP phones (such as a Cisco 7960).

You can actually call me from the Web using the following link under Internet Explorer. (Active-X based softphone…you’ll have to set your security to allow it to run.)

If you’re not using IE (personally I don’t use IE, or even Windows for that matter), you can download any SIP compliant softphone for your operating system of choice and call me directly by dialing sip://guest@mesquite.yi.org, just keep in mind this really will ring the phones in my house. You can also use a softphone on the freeworlddialup network.