As long as your dealing purely with SIP trunks and you're not trying to directly connect any analogue lines it should be fine. The jitter buffers at the endpoints sort out the timing problems in the hypervisor.
Any virtualization isn't recommended for VoIP since you don't have a real clock source and will at some point run on timing issues since you don't have a reliable timer (ofc that this gets worse depending on the load of the server and shouldn't be a big problem for someone running low call volume).