ZOOM ATA Busy Tone

When I dial a PSTN number via voip and the person is engaged I do not get a busy tone, it just rings a few times and then goes dead. This is using the ata. When using a softphone I do get a busy tone. How do I get a busy tone when using my ata, is it likely to be a setting on it somewhere?

Is it possibly because the call progress tone is wrong? Do you know the corect uk setting?

Also I cant get a 2nd voip account to work with a prefix. When I insert '2' as a prefix and then dial 2 and then the number the call doesn't seem to go out via my 2nd account. Thanks.

I've not noticed any problems with engaged tones on my zoom ATA; but I may have not called anyone who was engaged at the time. I sometimes get fast-busy signals (like a double-rate engaged tone) if the call cannot be completed for some reason.

For using multiple accounts, I think the name "dial prefix" is misleading, they're more accurately dial /patterns/. Here is mine; I have an account with Telasip for US incoming and outgoing, and an account with Voipfone for UK incoming and outgoing, and I want any number starting 1 to dial out as a US number, any number starting 0 to dial out as a UK number, and anything else to dial out as a local US number in my area code (919). To do that, I have the following dial prefixes:

Telasip / US:


This means - any number starting 1, route out on this account, any number starting 2-9 and 7 digits long, prepend 1919 and route out.

My voipfone / UK prefix is:


This means, any number starting 0 followed by 1-8 (i.e. anyhting but premium rate UK calls) route out on this account, anything starting 001, replace the 001 with 1 and route out (this I use for the voipfone shortcodes, e.g. for voicemail I dial 001571 rather than just 1571 which would get confused with a US number).

So, if you wanted to use a second account when you dialled 2, and you wanted to have to ADD the 2, rather than it be part of the number sent out to the second provider, you'd need something like:


If you have any questions, let me know and I'll try and answer them.

Thanks for the help. I can kind of see how it works now and had a go at it. Basically the prefix of 0 would mean anynumber begining with 0 will go out via that account. Not dial 0 this connects your to 2nd voip account then dial the number as I had thought. I would like to be able to put the same number such as 0044.... out via the account of my choice by dialing a prefix. Kind of like if I press #8 my call goes via pstn. I tried your recommendation but no joy. Any ideas?

Sorry, not quite sure what you mean...

If you want to replace some numbers in the dialed pattern with some others, the notation is:


Where xx is what you dial, and yy is what it gets replaced with...

Separate different patterns with a | character.

Does that help at all?

I have 2 voip accounts. When credit is low on the 1st account I would like to be able to use the second account by dialing a prefix.

So if i dial a number say 00441234 it just goes via my first voip account by default.

However today I have no credit on my 1st voip account so I need to use my 2nd account. I have specified <2> as the dial prefix. However when I dial
<2> 00441234 it does not work and I get an error tone or get routed to some foreign number. Thank you, sorry I did not explain myself better. :roll:

OK, for this to work then you need, for provider 1, some sort of pattern that matches as you normally dial, e.g. for UK numbers:


Then for the second provider:


This will take the prefix 2, strip it out, and dial out the number as dialled.

If you wanted to dial the full international number, then these two should work:

Provider 1:


Provider 2:


That will route otu any numbers starting 00 to the two providers...

hth some, let me know if you have any more questions.

You have opened the floodgates now :wink:

Is there a way to turn up the volume on normal PSTN calls particularly my voice going out, and also at the same time turn the volume on voip calls down a tad? I have tried various things but if one goes up or down the other seems to do the same.


Going on what the Linksys UK and US busy tones are I make the UK tone on the zoom to be

1 2 0 400 -19 375 375

However I still don't get a busy tone, either it sounds like it is ringing normally or there is nothing.

Will do. When I have a few minutes I intend to write up a page for voip-info.org about the ATA, with the things I've found out. It's a real mixed bag, 'cause the device works wonderfully and the sound quality is brilliant IMO, and the price is great too, but the documentation is truly appalling, you have to sit there and plug through things and make guesses to work out how to do anything.

Ah well! Can't have everything for the price they sell 'em for I guess. I think it's a false economy for them to not document the thing properly, the price point they're selling them at could make them incredibly popular if only they were at all easy to use...

Hello again. Still waiting for Zoom tech with regard to the busy tone. Doubt I will find a solution. No biggie.

Another question though. The ATA is supposed to support QoS. Do you know if it is configured for this by default, just that I am trying to set up my speedtouch 576 to send voip first but I cant seem to get it to work properly, it doent appear to be putting traffic in the real time queue. Calls tend to break up if I have any sort of P2P running and I am trying to stop this.

Who is online

Users browsing this forum: No registered users and 0 guests

Copyright 2004 - 2017, iNet Telecoms® Ltd. All rights reserved.