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

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.

_________________
Bus error (passengers dumped)

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...

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

As a follow-up to this, I am trying to get Asterisk set up on my Linode just for some messing around - at this point I just want to do some softphone-to-softphone calls.

Does anyone have any recommendations on a guide to getting started? I've got Asterisk installed and I've mucked with the basic sip.conf and extensions.conf files, but when I try to get my laptop to register with asterisk using SJphone it can't seem to authenticate.

I'm a total Asterisk novice, so I don't even really know how to start debugging things.

Thanks!

First thing is to get to the asterisk console. Easiest thing to do is 'asterisk -r' from the shell, which will connect you to a running asterisk instance.

Once you're at the asterisk '*cli>' prompt, 'core set verbose 5' will get a lot more info printed to the console about what asterisk is doing.

Other particularly useful commands are 'sip set debug on' and 'sip show registry'. I usually have 'qualify = on' in my sip.conf definitions, which will also give more info when using 'sip show peers'

There's built-in help, try commands like 'help sip' and 'help core' to get a little more info.

Hope that helps,

Paul

That's very useful - thanks.

I'm trying to connect to my server from behind an Apple Time Capsule. Would my local NAT be preventing me from talking to my Asterisk server properly, and if so, any suggestions on what I can about this?

Yes, NAT can definitely be a problem. There's some info on why here:

http://www.voip-info.org/wiki-NAT+and+VOIP

-- Paul

Woo! Got it working. The trick was adding

nat=route

to my sip.conf for those particular extensions. Thanks again!

Glad to hear it - now the fun can really start

By the way, when you progress to the point that you want to make calls to/from the PSTN, I've been getting good results with www.flowroute.com. Inexpensive and good quality, a rare combination. They're also one of the few voip providers that gives you per-call control over your outgoing caller ID.

Check out chan_mobile too - basically integrates bluetooth support into asterisk, so cellphones (and BT headsets) can be used. Not applicable if your asterisk is remote, but very cool if you run one at home instead/as well. You can detect if a particular phone is in range, and your dialplan can behave accordingly. And asterisk can make & receive calls through paired cellphones in range.