Linode Forum

[ShakataGaNai]

I'd like to run Asterisk on Linode, but I'm concerned with how well it will work and what it takes to run (out of the ordinary). I found the guide "Deploy VoIP Services with Asterisk and FreePBX on Ubuntu 9.10 (Karmic)" but that's from quite a few versions ago. Has anyone done anything more recently? Is it still a special setup? Is there a better guide?

[brandon711]

Hello,

I just finished a asterisk 1.8 setup with freepbx 2.9. It went really smoothly, and is working well about a week later. The only thing that is out of the ordinary is getting the DAHDI modules to compile. It was a little more of a challenge than normal as I used debian, and freepbx doesn't completely like that without a little work.

In the past I have managed to get trixbox running on linode by setting up a web repo and booting with pv-grub. This would probably work for the current distro's as well.

I didn't use any guide for this as I have done many asterisk installs, If you can give me a bit more info on the configuration your wanting I may be able to give you some pointers.

~Brandon

ShakataGaNai wrote: I'd like to run Asterisk on Linode, but I'm concerned with how well it will work and what it takes to run (out of the ordinary). I found the guide "Deploy VoIP Services with Asterisk and FreePBX on Ubuntu 9.10 (Karmic)" but that's from quite a few versions ago. Has anyone done anything more recently? Is it still a special setup? Is there a better guide?

[ShakataGaNai]

brandon711 wrote: Hello,

I didn't use any guide for this as I have done many asterisk installs, If you can give me a bit more info on the configuration your wanting I may be able to give you some pointers.

~Brandon



Much appreciated. I was planning on using Ubuntu 11.04 and going with the repo package (1.8.9 I think). I've installed Asterisk a number of times, but never on a VM/VPS. I was planning to run with something like Flowroute for SIP trunk, and some Cisco SIP phones connecting remotely.

[brandon711]

If your planning on using meetme you will need to compile the dahdi modules, this can either be done against the kernel source using Linodes kernels, or you can use pv-grub to run the distro's kernels. I used PV-grub.

It should work out well for you. I am using the repo from digium for asterisk on debian.

Are you planning to use freepbx or all manual configs?

[ShakataGaNai]

brandon711 wrote: If your planning on using meetme you will need to compile the dahdi modules, this can either be done against the kernel source using Linodes kernels, or you can use pv-grub to run the distro's kernels. I used PV-grub.

Ok, Good to know. I'm going to try using ConfBridge instead of MeetMe since that's the new hotness (at least in A10). Since that's written into core asterisk, it does't need DADHI (at least that is my understanding).

brandon711 wrote: Are you planning to use freepbx or all manual configs?

All Manual Baby! (Sadly I've done this enough times that I basically think in extension.conf terms everytime I call someone else with a phone system)

[brandon711]

Its not that I don't understand the extensions.conf file. Its that I have to build systems that the end user can modify, and until I write my own webgui. its freepbx for most systems I make.

I have not done anything with 10. yet. Good luck, I don't think you will have any problems on here. We have had really good luck so far. :)

Let me know how it goes.

ShakataGaNai wrote:
All Manual Baby! (Sadly I've done this enough times that I basically think in extension.conf terms everytime I call someone else with a phone system)