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" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> 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 > /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE > / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
