Category Archives: Asterisk and VOIP

Asterisk on an FPGA

From the Beer, Coffee, and a little DSP » Blog Archive » Asterisk on an FPGA

Over the past couple of years a few people have suggested running Asterisk on an FPGA using an embedded processor core. I must admit I had always assumed that the processor would be too slow to be useful, certainly much slower than a regular embedded processor at the same price.

However my friend Stelios Koroneos and the team at Digital OPSiS have proved me wrong! They have managed to implement Asterisk on a Xilinx Virtex 4 FPGA, running a 300MHz Power PC core. These FPGAs cost about the same as an embedded processor, e.g. around $12 in Qty 1000.

I’ve been interested in FPGAs and Asterisk: it’s good to see projects that do both.

Help me with a VOIP experiment…

I’ve been experimenting with voice-over-IP telephony using the Asterisk open source PBX system, and I’m at the point where I’d like you to help! I’ve setup a phone number with VoicePulse Connect! to route to my Asterisk server, and configured a very simple extension so that you can record comments about my blog. Whatever charges you normally acrue will be charged for this call, but feel free to waste some of those free nights and weekend minutes and a minute of your time to try it out:

When you dial 510-323-0224, you’ll hear this:

Welcome to the brainwagon.org comment line, powered by Asterisk, the open source PBX system. I’m experimenting with using VOIP technology as part of my weblogging and podcasting experiments, so I’d appreciate it if you’d go ahead and leave a message at the beep. Feel free to include any comments (positive or negative) about my blog and please include a brief rating of the voice quality (say, from 1-10) and whether you accessed this number from a cell phone, a regular land line, or via some other VOIP service like Vonage. Your comments will be recorded, and I reserve the rights to retransmit them as part of a future podcast. If that’s okay with you, wait for the beep and record your message. When you are done, hangup, or you can hit the pound sign, and it will play back.

Ultimately I’m interested in using Asterisk to setup telephone conferencing with automatic recording so that I could do a podcast with others, all linked together via phones of any convenient sort. This is one small step in that direction. Thanks for participating in my experiment.

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VoIP for free with the Sipura

According to my UPS tracking code, my Sipura SPA-3000 should be awaiting me when I arrive home today, and in a timely fashion, I found this link for setting one up to do free/cheap VOIP:

GRYNX » VoIP for free with the Sipura

I’ll post a complete review of the device itself after I’ve tinkered with it for the weekend.

Incidently, if you are using Firefox and Greasemonkey, you can use this script to turn UPS and Fedex tracking numbers into clickable links that go directly to the right page on their respective service for tracking.

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What I know and don’t know about Asterisk…

Well, I haven’t got it all figured out yet, but here is what I know:

  1. It’s not hard to compile asterisk for the amd64, but…
  2. You need to be careful compiling the ztdummy kernel module: in particular, you must define USE_RTC when compiling, because the sourcefile incorrectly only checks for USE_RTC if __i386__ is defined, which is not the case for 64 bit machines.
  3. Getting mpg123 to compile seems hopeless on the amd64.
  4. You can get a phone number for $11/month, with unlimited incoming calls and about $.02 a minute for outgoing calls to the United States via VoicePulse Connect!.
  5. I can, with a minimum of effort, create a simple dialplan that allows incoming calls from that number to be forwarded to some phones attached to my hacked Linksys PAP2 adaptor.
  6. BestBuy has a nice 5.8 ghz phone on sale for $25 with a $10 rebate.
  7. I can make calls from my Linux console (using a headset) to those phones.

What I don’t know:

  1. While the phones attached to the PAP2 can receive calls, I haven’t figured out how originate calls from them.  There is obviously something stupid about configuring them that I don’t understand.
  2. Sometimes the voice quality from the console seems bad, which I suspect is some kind of codec mismatch.

Ultimately, I’d like to make a simple “dialplan compiler” that allows you to enter some basic information, (account numbers, logins and the like) and generates the necessary asterisk configuration files.  I’ll probably write it in Python…

Hopefully, I’ll have this up and ready for human testing in the next week or so.  Then, I’ll unveil the deeper purpose!  Stay tuned.

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