My review of the AT-530 did not go unnoticed, particularly one of my critical comments regarding the display which was permanently set to ‘VOIP Phone’ – not particularly useful I’m sure you’ll agree. Well, a couple of days ago an email arrived with a rather large attachment – a new firmware update hot off the Atcom presses (now also available on the Atcom website download area).
The new firmware offers a few new features:
Version 07-04-24 Release Note:
1.Add eight different ring tone
2.synchronize from SNPT server
3.Add the sliding bar of menu setting page
4.Add LCD logo display customization function
5.Display the current server type ( SIP or IAX)
You’ll note No. 4, which I can confirm works. I have yet to check the other new features (keep an eye out for the full review soon), but this new firmware does appear to have broken one previously working function. Now, if you set the time zone to GMT then you just cannot enable daylight savings, so the clock will be an hour slow during the summer months. Back to the drawing board guys !
May 8, 2007 at 1:15 pm
[...] 2: I have written a follow-up to this review here. Posted in Telephony, [...]
May 11, 2007 at 4:30 pm
How did you change the display?, I cant seem to find a setting for it anywhere.
May 11, 2007 at 4:45 pm
Hi John,
Go to the ‘System Mange’ section, then click on ‘Account Manage’. Near the top on the right-hand side there is a field called ‘Lcd Logo’. Stick your display text in here. I recall having problems initially with the text appearing on the phone’s display, but I think a re-boot sorted that.
I’d be interested in hearing your opinion of the phone overall.
Cheers,
Colman
May 11, 2007 at 5:05 pm
Aha, Account Manage, what a weird place to put it… Ring Type is more obvious in Audio Settings. I’ve been having a devil of a time with echo today since plugging the phone in, but between my atcom x100p clone being on FCC mode and wrong opermode little wonder! (I’m in the UK) I’m a little confused as to how I answer a second call while on the the phone already as the phone rings but no other prompts appear for me. As you have said already, the manual is a bit light on detail with regard to waiting indicators and such, also a nuisance is the habit of chopping off the beginning of the CID on calls. My trixbox CID-prefixes are chopped, which has forced me to use Agent prompts on the lines… Anything else I find I’ll come back with.
Best Regards
May 14, 2007 at 8:59 am
John,
I’ve being using the phone fine with no echo problems…I’m in the UK too. Having said that my Asterisk box is not hooked up to the PSTN at the moment, that may be a reason ?
When a second call comes in press the ‘Hold’ button and you will put the first call on hold and pick up the second one. Pressing ‘Hold’ again will hold the second and connect you back to the first.
Can you elaborate on the CID-prefixes issue…I haven’t had any problems with them.
Cheers,
Colman
May 23, 2007 at 3:58 am
Hi fourlakes,
I have just setup a new Asterisk server in my friend’s company. We have opted to use AT-530 to works with our new IP-PBX.
With the latest fireware, everything seems OK but with one serious problem encountered:
If A is the receptionist and is using AT-530, B calls A from PSTN and A answer it. No problem here.
Then C calls A from PSTN, A got the call-waiting signal and press the “HOLD” button to switch from B to C. When A talks to C and knows the call should be transferred to another colleague by pressing “FWD” + Extension + “#”.
As a result, A got disconnected and B can talk to C directly.
Actually, the receptionist wants to transfer C’s call to another colleague and then talk to B again. But the outcome gets A disconnected.
I try to abandon the AT-530 transferring capability and use the one provided by Asterisk instead. However, when there are two calls answered by AT-530, no matter what button I pressed, the respective DTMF cannot reach the Asterisk server and hence cannot get the transfer done.
I am still figuring how to solve the problem encountered.
Cheers,
Kenny
July 16, 2007 at 7:57 am
The Atcom AT-530 is a nice entry-level phone.
The built-in switch / NAT router means it can adapt to various Internet connection environments.
The IAX support (hard to come by in a hardphone) is very welcome, however I’m not sure if there’s a way to switch between IAX and SIP for outgoing calls on a call-by-call basis.
Some things that I am hoping Atcom will add to the firmware are:
1. ENUM lookups – so the phone will use DNS to automatically check e164.org, e164.arpa, etc, to find free SIP or IAX addresses for the phone numbers I dial.
(This could be done directly using DNS, or via the “SipBroker” web service:
http://www.sipbroker.com/sipbroker/action/webServices
)
2. Make the firmware and/or SIP stack a bit more tolerant of SIP proxies that don’t add CR LF, as explained here:
http://forum.voxalot.com/voxalot-support/53-direct-dial-ip-2.html#post6823
3. Make the NAT router support “stealth” mode, so it ‘asses the grc.com test
September 13, 2007 at 9:24 am
[...] far as I can see, there are no new features, although a previously reported issue with the Daylight Savings setting does now appear to have been resolved. However, the update [...]
February 29, 2008 at 6:06 pm
Hi fourlakes,
I’ve tried to upgrade the AT-530 but I had no luck. I’ve used ftp, TFPT and HTTP, however the phone shows Failed message in evry trying. Any suggestion??? Do you know if the new firmware eliminate de missing calls message from screen?? Thanks a lot
Martin
February 29, 2008 at 8:59 pm
Martin,
Sorry to hear you’re having problems…I upgraded to V 07-09-14 pretty easily using HTTP. I presume you are on the same subnet as the phone ? Which firmware are you trying to upgrade from ?
No joy with the ‘missing calls’ message I’m afraid, still get it. Unless there’s a configuration option I’ve missed somewhere. To be honest, it’s pretty stable in other areas and I’ve learned to live with this particular foible by dialling *97 to check my voicemail if I’ve missed calls. Setting Asterisk up to email voicemails to be helps too
Keep us posted on your progress,
Colman
March 6, 2008 at 6:16 pm
Thank you Fourlakes
Well this is the problem: We have connected 8 phones to a VoIP Central. What I want is that the phone doesn’t count the missed calls because for incoming calls there are 3 phones that ring (A, B and C). If A or B answer the call, C registers a missed call. At the end of the day, each phone has more than 80 missed calls registered and the way to erased is very annoing (you have to erase thm one by one).
If you have a way to stop the counting of the missed calls would be great.
Best wishes,
Martin.
April 20, 2008 at 6:29 am
Hi fourlakes,
You able try the VPN feature of the phone. I’m particularly interested in secure communication between remote AT 530 and Asterisk.
Scenario :
AT 530 establish VPN connection with the central office VPN Server and login to office network . Then use local network ip addresses to establish connection between AT 530 and Asterisk.
thanks
chamara
April 21, 2008 at 8:18 pm
Chamara,
Sorry, I haven’t tried the VPN feature of the AT-530 yet….I doubt I’ll get the chance to do so too. Sorry.
Colman
May 22, 2008 at 6:09 pm
hello,
please am new to sip and trying to configure the at-530 ith 3cx and i am having a hell of a time. Please is there anywhere i can get a step by step manual for configurating the at-530.
May 29, 2008 at 4:03 pm
I am currently trying to connect to the FreeWorld iax service (to punch through a problematical firewall – to which i don’t have access). I wonder if anyone has done this yet? FWD’s information is famously lightweight, and assumes that you are connecting to them from an Asterisk server. It looks easy from the web setup page for iax, but the phone just will not connect.
June 21, 2008 at 5:14 am
hey there,
I am planning mass deployment of AT 530 phones,instead of uploading cfg files manually, i would like to use a TFTP server together with option 66 on DHCP server. What is the file format to be placed in the TFTP server? is it
.cfg or any other? Do you happen to know the syntax in that file??
Thanks for your help.
September 27, 2008 at 3:00 am
[...] It’s important to remember that having a stable and reliable high speed internet connection is necessary to establish and maintain video conferencing. Using a high-speed internet connection is preferable that way the experience not only workable but also enjoyable [...]