After running PBX in a Flash (PiaF) in a virtual machine for a few days, I had a couple issues:

The system would generate an occasional message "TSC appears to be running slowly. Marking it as unstable." This issue did not keep the PBX from running.

Voice quality was sometimes jittery. In a call, it would sometimes "cut out" (like a bad cell connection). In voicemail, even listening to the pre-recorded prompts, there was frequently some garbled-sounding audio.

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In my original PBX in a Flash setup, Asterisk is set to "listen in" on calls. One advantage of this is that it can then respond to commands sent from a phone keypad during the call, e.g. to transfer or record the call. However, it does add processing overhead. If your devices support it, you can tell Asterisk to butt out of your conversations, allowing connections to "bridge" directly between endpoints and avoid going through Asterisk. Here's how.

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I want to use a true SIP phone for my office phone. I was initially attracted to the Polycom SoundPoint IP 430 because of its support for electronic hookflash with my Jabra GN9350 wireless headset. By using a special adapter cable, I can answer the phone from the wireless headset without the need for a mechanical handset lifter.

Things I Wish I'd Known

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I've just posted a series of articles on selecting a VoIP PBX and setting it up under Microsoft Virtual Server. To read the series from the beginning, see Getting Into VoIP.

Setting up an Analog Terminal Adapter (ATA) for a VoIP system is normally something you do while configuring the PBX. However, since it is configured as a separate device using its own web interface, it seemed to warrant its own blog entry.

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If you've been following along through the introduction, part 1, and part 2, you now have a PBX in a Flash (PiaF) setup running under Microsoft Virtual Server. It's keeping time, it knows who it is and can send out mail. Now it is finally time to configure the PBX itself!

Setting up the PBX involves a few steps:

Updating and Adding FreePBX modules
Configuring Trunks, Extensions, and Routes
Configuring Other Features

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The first article in this series, Getting Into VoIP, gave an overview of VoIP systems that I tested and why I chose to implement PBX in a Flash (PiaF). The next post, Setting Up PBX in a Flash, Part 1: Configuring a Virtual Machine talked about setting up PiaF under Microsoft Virtual Server. In this post, we'll finish all the steps needed to complete our basic installation before configuring the PBX itself.

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The introduction to this series, Getting Into VoIP, explained some of the requirements and testing that went into choosing a VoIP environment and application. One really-want (if not must-have) requirement is getting it to run in a virtual machine. There are so many advantages to virtualization:  easy backups and moving VMs to different hosts, to name two. However...

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It all started with my new Magicjack. Plug this little gizmo into your PC and you get unlimited domestic calling, $40 for the first year and $20/year thereafter. That's hard to pass up. But it also means that I now have three phone lines coming into the home office. So I either needed to get a bigger analog phone, or get into a PBX.

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