alexd wrote:
[color=blue]
> Dan wrote:
>[color=green]
> > 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=green]
> > 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=green]
> > 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=green]
> > 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].[/color]


Thanks very much for your help... I did what you suggested and brought
it back inside the network... it was a NAT issue, and the polycom
handsets don't support keep-alive! My good chap Ray sorted it with a
little bit of beer and a lot of shouting. Works like a treat now!

Thanks very much again,

Dan