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Dear All,

Really simple setup, no manual conf file changes, just a box running
trixbox with two extensions (Polycom 301s) using NAT (done by
FireBrick) - no connection to the outside world, just internal calls
for now.

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.
Happens the same if softphones are used (SJPhone on OSX). Call
durations are reporting correctly.

Any ideas? Everything has gone so smoothly (so far).

I would be really greatful for your advice on this one - two of us have
speant all day on it without so much as an inch of progress!

Thanks,

Dan

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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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