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Does anybody have experience or running Asterisk on VMware(?) - specifically the latest ESX...?

I've got a proof of concept working but am reticent to roll it out without a bit more reassurance! I keep hearing about problems with timing...?

Anybody done it?

Mike

  • Hey guys - thanks for all your advice. Just a note to say we put this in now about a month ago and (touch wood) it's all working just fine. No chop, no dropped calls, no weirdness. That's another box turned off in the office in favour of virtualisation... Mike – Mike McClelland Nov 1 '09 at 20:04
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It works but there are applications that depend on an accurate clock that need some special attention, for example conference bridging. A good test is to install asterisk + zaptel and compose a test dialplan for a conference bridge. Connect to the bridge with two phones and listen to the audio.

  • Cheers Lee, I'll give that a go. I have to say that I have already set it up now and the basic system tests are all fine (e.g. Echo Test, etc.) - however, I hadn't tried the a conference bridge. Will give it a go, thanks! Mike – Mike McClelland Aug 30 '09 at 17:56
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Several years ago we ran it in a VM on a basic Pentium 4 host - and it was no good. Voice was choppy and cut up. We installed it natively on the P4 (I think it was a 1.6) and it was all good.

Since then we haven't installed it on any VMs, but I've heard from others that just as long as you assign it enough resources it should be fine.

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I've run it inside ESXi 3.5 on a BladeCenter server without issues; the main source of issues tends to be the need for a very reliable timer. The nature of virtual machines kinda goes against consistent software timing sources, but with Linux 2.6 and VMware ESXi, FOR ME, it worked well. YMMV for Hyper-V or Xen or VirtualBox.

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I run an asterisk PBX on my Mac through Parallels. It works great, tested only with two phones and going to the PTSN using IAX routes. This virtual machine its for testing and development proposes. With those two phones I tested conferencing just fine, the clock source was of source ztdummy but it worked ok. As long as you have enough RAM and CPU available it should work ok.

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It used to be that extra care was required when virtualizing asterisk however more recent versions support reliable timing that is not based on dahdi or zaptel. You can see those timing options here: https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces. So virtualization is a great option. I use this setup all the time to provide high-availability in conjunction with a sip based analog gateway for pstn connectivity

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