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By WelshPaul
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Purchased a Grandstream telephone adaptor?
If you have purchased a Grandstream telephone adaptor (ATA), it will come configured for the US telecom system as standard. As such, it will play back US style tones, sounds and some UK phone features may not work correctly unless you make some configuration changes via the devices web based GUI.

What are the UK regional settings for the Grandstream telephone adaptors and how do I implement them?
Login to your Grandstream telephone adaptors web based GUI by typing the IP address of the device into your web browser. (There is no need to enter "http://" before the IP address.) If prompted enter the username and password to complete the process, you should now have access to the devices Web based GUI, from here you can configure your device with the correct regional configuration settings for use in the UK.

As an example, on the Grandstream HT812 configure with the following:
  • Navigate to the BASIC SETTINGS page:
    • Self-Defined Time Zone: GMT0BST,M3.5.0/1,M10.5.0
  • Navigate to the ADVANCED SETTINGS page:
    • System Ring Cadence: c=400/200-400/2000;
    • Dial Tone: f1=350@-19,f2=440@-22,c=0/0;
    • Ringback Tone: f1=400@-20,f2=450@-20,c=400/200-400/2000;
    • Busy Tone: f1=400@-20,c=375/375;
    • Reorder Tone: f1=400@-20,c=400/350-225/525-0/0;
    • Call Waiting Tone: f1=440@-20,c=300/10000-300/10000-0/0;
    • Prompt Tone: f1=520@-19,f2=620@-22,c=0/0;
    • NTP Server:
  • Navigate to the PROFILE 1/2 (FXS PORT on HT813) page(s):
    • Dial Plan: { 100 | 101 | 111 | 112 | 155 | 195 | 999 | 116xxx | 116111 | 116123 | 118xxx | 1471 | 157[1-2] | 08001111 | 0845464x | 0[1235789]xxxxxxxxx | 00xxx. | x+ | \+x+ | *x+ | *xx*x+ }
    • SLIC Setting: uk
    • Caller ID Scheme: sin 227 - bt
    • Hook Flash Timing: Minimum: 60 Maximum: 200
    • Ring Frequency: 25
    • Ring Tone 1: c=400/200-400/2000;
    • Ring Tone 2: c=400/200-400/2000;
    • Ring Tone 3: c=400/200-400/2000;
    • Ring Tone 4: c=400/200-400/2000;
    • Ring Tone 5: c=400/200-400/2000;
    • Ring Tone 6: c=400/200-400/2000;
    • Ring Tone 7: c=400/200-400/2000;
    • Ring Tone 8: c=400/200-400/2000;
    • Ring Tone 9: c=400/200-400/2000;
    • Ring Tone 10: c=400/200-400/2000;
Remember to click on the "Update" and "Apply" buttons located at the bottom of every page to save and activate the changes.

Purchased a Grandstream HT813 telephone adaptor?
The HT813 is an analog telephone adapter that features 1 analog telephone FXS port and 1 PSTN line FXO port in order to offer backup lifeline support using a PSTN line. Additional UK regional settings are required for this model and I have included them below.

Navigate to the FXO PORT page:
  • Caller ID Scheme: SIN 227 - BT
  • FSK Caller ID Seizure Bits: 96
  • FSK Caller ID Mark Bits: 55
  • PSTN Disconnect Tone: f1=400@-30,f2=400@-30,c=0/0;
  • Country-based: UK
  • Impedance-based: COMPLEX3 -- 370 ohms + (620 ohms || 310nF)
Grandstream HT813 UK Dial Plan:
Code:ย Select all
{ L: 10[01] | L: 11[12] | L: 195 | L: 999 | L: 147[457-9] | L: 17070 | 155 | 116xxx | 116111 | 116123 | 118xxx | 1471 | 157[1-2] | 08001111 | 0845464x | 0[1235789]xxxxxxxxx | 00xxx. | x+ | \+x+ | *x+ | *xx*x+ }
More information about the above Grandstream HT813 UK Dial Plan can be read here: @UKVoIPForums @ Grandstream HT813 UK Dial Plan
CJT-80, andrewclark55 and 1 others liked this
Thank you @golightlygl for your kind words and welcome to the forums. ๐Ÿ‘‹๐Ÿป๐Ÿ‘

FYI - I have updated my original post to include additional settings that are needed when using a Grandstream HT813 telephone adapter. ๐Ÿ˜€
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By martinwinlow
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Holy mollies... and I thought configuring a router was non-user-friendly.

There must be at least a hundred different things one can tweak with this thing... and clearly I have not enough basic understanding of what is needed to get mine working.

My VOiP provider, A&A, appear to be under the impression that the only people who will ever need VOiP (and therefore need to set it up) are tech-wizards as it is like pulling teeth trying to get even basic configuration settings and instructions out of them.

Not really ready for the masses...
andrewclark55 liked this
Andrews & Arnold are one of the best reviewed VoIP ITSP's around and are no more difficult to setup than any other VoIP provider. ๐Ÿ™‚

Check out their support site if you need help configuring a device (feel free to ask here too):

About the settings posted above by @WelshPaul, well they are not required but are recommended. These devices are pre-set for use within the North American market and so long as you don't mind your phone tones or ringer sounding different then you can safely ignore them. ๐Ÿ‘
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By martinwinlow
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Thanks for your reply.

A&A have provided 5 'generic settings': Registrar:
Proxy: Leave blank
Authentication Username: Your Number in International format, e.g. +441234567890
Password: Your SIP Password

... unfortunately none of the hundreds of settings on the Grandstream HT801 ATA I I have match these. Take SIP Password for example. The Basic and Advanced settings page for my ATA don't mention it at all, it is only on the last 'FSX Port' page that you find 'SIP'... and 'only' *37* times!

Asking a layperson to interpret what A&A provide and apply them to their own particular ATA is simply ridiculously unrealistic. I think it would pay them (and any other forward thinking VOiP provider who wants to cash in on people moving over to VOiP - we all will eventually!) to supply basic functionality settings *for all the most common ATAs* - or even a selection of them. I guess all this answers the question of whether or not I should have bought one of A&A's ATAs!
Hi @martinwinlow, give the following a try:

  • Account Active: Yes
  • Primary SIP Server:
  • Outbound Proxy:
  • NAT Traversal: Keep-Alive
  • SIP User ID: Your Number in International format, e.g. +441234567890
  • Authenticate ID: Your Number in International format, e.g. +441234567890
  • Authenticate Password: Your SIP Password
Hopefully that is all that is required in order to get your Grandstream HT801 registered to your A&A account. ๐Ÿคž
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By martinwinlow
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Thanks but I didn't see your reply until just now but I have spent the last 4 hours or so to-ing and fro-ing emails with A&A whose last email reads...

We supply Snoms and Gigasets which we'd be happy to help you with, and
usually go out preconfigured, plug & play.
However you chose to provide your own hardware, which you are entirely
entitled to, and we are more than happy to supply the details with which
to set in the configuration.
We don't have direct experience of configuring Grandstreams but plenty
of our VoIP customers use them.
Different manufacturers of hardware and software use different names for
thew same thing.
At a guess 'Primary SIP server' is where to put
'Prefer Primary SIP server' I would guess you set to yes.
'SIP user ID' is your number in international format.
'AUTH ID' same as SIP user ID I'd guess.
'Authentication Password' should be the SIP password on for your number.
If it's connected that should be enough to get a registration, as I said
we haven't yet seen a registration attempt from your hardware reach us yet."

Having applied those 4 parameters, the line now works nominally and shows 'registered' in the ATA's Status page (contrary to A&A's statement).

I reiterate, how any ordinary soul would figure out how to get their HT801 (or probably any other ATA) working without any help is a total mystery.
Andrews & Arnold support clearly isn't as good as I thought it was if it took four hours to resolve such a simple support request. Poor spelling, grammar and the fact they use of the word "guess" in multiple examples doesn't fill one with confidence!
martinwinlow wrote:I reiterate, how any ordinary soul would figure out how to get their HT801 (or probably any other ATA) working without any help is a total mystery.
Most VoIP providers sell pre-configured devices that only require the user connect them to a power source and internet connection in order to make and receive calls and thus they don't need to figure anything out. Sure these cost a little more but you're paying to have a professional do all the hard work for you! The option to use your own hardware is great for users who require custom configurations and advanced features. ๐Ÿ‘
@dcowan On both SIP and PSTN or just PSTN?

Who supplies your PSTN line? Some telco's require a CLI subscription. If CLI is enabled on the line, have you tried using a different telephone? After all, it could be a fault with the phone and not the HT813 or PSTN line.
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By andrewclark55
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Thanks Paul for taking time to post all that info.

I have entered all of it with my new HT801 on a new Sipgate account connected to a Draytek Vigor 2862ac router but without success after trying setting various other suggestions from the Grandstream forum, including updating the firmware and setting up an entry for a STUN server.

Symptoms are I can dial out OK and complete calls but incoming calls don't ring. Initially I was still able to answer my own test calls made from my BT line although it didn't ring but now, after trying various settings changes I find that calling the VOIP number just results in hearing one ring on the calling phone followed by the unobtainable tone.

Can anyone throw some light on this issue or point me to a configuration file that will actually work with Sipgate (not the screenshot they give as I tried that and already had to make changes to what they said which seems to be for the US market).

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