Dan wrote:
[color=blue]
> Dear All,
>
> Really simple setup, no manual conf file changes, just a box running
> trixbox with two extensions (Polycom 301s)[/color]

Ah yes, the perpetual beta that is Trixbox ;-) Which version of Trixbox? Are
your Polycoms on the latest firmware?
[color=blue]
> using NAT (done by FireBrick) - no connection to the outside world, just
> internal calls for now.[/color]

Are you saying that the FireBrick is doing NAT between the Polycoms and the
Asterisk box? If that's the case, I'd eliminate the NAT first and confirm
it works without that.
[color=blue]
> All works fine for first call in each session, with the phone
> registering as available. However, once a call has been made, the phone
> becomes unavailable and all subsiquent calls to that extension getting
> shoved to voicemail (busy) and the status reporting as unavailable. It
> is like the phone system doesn't know that the call has terminated.[/color]

What does the Flash Operator Panel say? [although on 1.1.1 FOP server seems
to need restarting on a regular basis as it gets out of sync].
[color=blue]
> Happens the same if softphones are used (SJPhone on OSX). Call
> durations are reporting correctly.
>
> Any ideas?[/color]

Log into an asterisk shell with 'asterisk -rc'. Then type
'sip show channels'. This will show you any active SIP calls [I presume
you're using SIP].

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