Linode.com Forum

[PaulC]

I've done a quick search for 'Asterisk' on the forums, but didn't find much recent discussion (specifically post-Xen migration).

I have asterisk 1.6 already set up on my home machine (used when my parents to call us from the UK), but then realized my linode uptime has been better than my residential broadband connection... (which is usually great, but occasionally goes through rough periods).

We're planning to ditch our landline entirely, and port the number to a VoIP provider (vitelity and flowroute look good at this point). I'd like to be able to 'fail over' to our cellphones if our internet connection is down. So I was thinking of running asterisk on my linode, to do some call routing and filtering.

I've also looked at various flavors of hosted solutions, although my ability to 'tinker' would be greatly reduced (my wife would see that as an advantage ;) )

Any words of wisdom?

Paul

[rjp]

If you don't need MeetMe conferencing, you should be able to get along fine. If you do, forget about it. I tried compiling ztdummy for conferencing support and installed it, but I quickly found that I could not successfully get a conference of more than two participants, and even that was iffy.

I don't think there's any VPS in existence that can provide the precise timing needed to successfully run MeetMe... you'd need a dedicated server for that.

[PaulC]

Thanks, rjp. No, I won't need conferencing. This is just 'normal' home/telecommuting use.

I don't think my linode will need to handle media streams (i.e. act as an STUN or media gateway), as I have a static IP at home and full control over the firewall rules. So I'm thinking the Asterisk instance on the linode will just be doing call routing, and some voicemail duties. Enough to make sure either our cellphones ring, and/or voicemails are taken, if our internet link goes down. And pre-empt the telemarketers, of course :)

BTW, I didn't need to install any zaptel stuff for asterisk 1.6 (either at home or on my linode, both Fedora 9). Perhaps they've finally removed the dependency? before that I had been using CallWeaver (fork of asterisk, which doesn't have a zaptel dependency either). The downside is that 1.6 is still beta, and it's different enough that some of that asterisk ecosystem advantage is lost.

I also took a look at FreeSwitch, since it's supposedly lighter weight on resources. But the documentation's too sparse for me to develop a warm fuzzy. Plus the config files are in XML...

[nessenj]

I've had no problems running asterisk on an Ubuntu installation (was as simple as doing apt-get install asterisk).

I am only running a Cisco 7940 on my desk that I use for work calls.

Jim