[Asterisk-Users] Re: SIP Phones-Receptionist Setup

[Craig Guy]

About once an hour the phone displays '403' on the display for about 10
seconds or so with this firmware.  There is no corresponding entry on the *
console.  'Spose it has something to do with registration.  Apart from that
it looks ok so far and the web interface now looks much better.

Craig

----- Original Message ----- 
From: "Mark Elkins" <mje at posix.co.za>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Friday, November 26, 2004 4:13 PM
Subject: Re: [Asterisk-Users] Re: SIP Phones-Receptionist Setup


> On Fri, 2004-11-26 at 09:05 +0100, hhandresen wrote:
> > OT:
>
> http://www.grandstream.com/BETATEST/
> (as someone else on this list stated)
> I've not seen any problems with it yet....
>
> Sequence is, you have a call, push Flash, dial new extension - speak,
> push transfer - and you're out of the loop.
>
> > But where did you get the 1.0.5.18 firmware ?
>
> > > PS - My Grandstream phones (BT100) with 1.0.5.18,
> > >      and Send-Flash-Event-as-DTMF=No,
> > > now are doing Attended transfer just fine!
>
> -- 
>   .  .     ___. .__      Posix Systems - Sth Africa.  e.164 VOIP ready
>  /| /|       / /__       mje at posix.co.za  -  Mark J Elkins, Cisco CCIE
> / |/ |ARK \_/ /__ LKINS  Tel: +27 12 807 0590  Cell: +27 82 601 0496
>
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