I’m loathed to put this as Part 1 or anything because then I’ll be obliged to make Parts 2, 3 and n.

Well I’ve been faffing with Asterisk for months now, slowly going nowhere, mainly due to having little or no time to put into it at work when ‘real work’ needs to be done, not having the time or the environment to concentrate on research properly or absorbing the docs, then having phones that don’t work, then having phones that work some of the time and well blah, blah.

About a month or so ago I had a breakthrough when phones actually started calling each other, which, stupid as it sounds took me ages to get right, what with never having used Asterisk before meaning I didn’t know when something looked wrong. Re-flashing the phones meant they picked up the server and then finding a commented out extension in a config file, even though I had added it using the web interface (everytime I added a new extension it would uncomment the previous extension and comment out the new one…).

Well yesteday, having been playing with incoming and outgoing calls via the PSTN using a Digium TDM-400 card, something didn’t seem right and, again, never having seen a working Asterisk box, I didn’t know whether some of the errors I was seeing were normal or not. It was only when I started playing with using the zaptel configuration tools that I realised that the error I was getting didn’t mean what I thought and why the thing wouldn’t dial out was due to the fact that the machine I was using wasn’t recognising the Digium card properly. A quick lspci revealed this to be true, I have no idea why, but possibly because I was using a cheap test machine with an SiS chipset as the Digium card wouldn’t fit in the case properly of the server I was going to use originally.

Well anyway, I wasn’t going to use this machine as a production server anyway, it was a proof of concept machine to see if Asterisk could do what we wanted (nothing that complicated to be honest). So, already sold on the idea, I made a fresh install of the latest Asterisk@Home (yeah I’m a wuss) on some proper server hardware and heh, I had it up in an hour or so, secured it and played with the PSTN again today. A bit of playing with the trunks and incoming and outgoing routes and I made my first calls in and out. I was really pleased to have it working, I left myself stupid smug answerphone messages.

The configuration is primitive at the moment, all calls out go over the PSTN and all incoming calls ring all extensions, but I guess thats to be expected. What I need to do now is assess call charges from BT and various SIP providers, analyse our call patterns and costs (mobile/international/regional) and then design a proper call plan. I also need to design a proper menu system and Digitial Receptionist so that people can get put through to the right person first time, sort out a default office voicemail, remote extensions over VPNs and arrange various out of hours and out of office forwarding to real home or mobile phones. This will be more evolutionary according to need, rather than revolutionary by designing everything from the beginning I suspect, but some planning won’t hurt at all, grumble…

I think thats about it on this subject, speak soon.