atcomsystems.ca/forum Forums TDM Business Telephone Systems Nortel Telephones & Systems DTMF tones through an ATA

Good call BCM

My bad - it may not be a problem with the ATA after all; I just verified that DTMF is not passed from one phone to another when both ends are Meridian phones.

So, is it possible to configure the KSU to pass DTMF tones during a call?

Thanks everyone.

That's what BCM was referring to while your on the call use Feature 808 (long tones)

feh, I see in your profile that you are a software engineer. If you are trying to develop a whiz-bang application for the Norstar, I understand if you are reluctant to share what you are trying to do. But, there is a lot of expertise here. And, the more information we have, the better we may be able to help you.

Don't sweat it Jimmy! I don't think we know what 'exactly' he wants just yet.

I found what I needed in the KSU guide; I had to configure the station to "receive tones". Once I did that, all was good.

Thanks everybody.

:thumb:

Sorry to drag out an old thread, but this is exactly the problem I am having and I would like to know what feh had to do to solve it.
In depth, I am attempting to interface a VOIP gateway/server to my Nortel Norstar PBX to enable me to add WIFI SIP phones to our site. I have a few ATA2's to provide analog lines, and I am having two problems with them.
1) As with feh, the DTMF signal is NOT getting through. When I attach an old school analog phone to the ouput on the ATA2 and call it from a nortel phone, there is not audible DTMF. I have heard about feature 808 and "long tones" but I am confused. Do I have to dial "feature 808" before EVERY call that I want dtmf to get through? Is there no better way?
2)The ATA2 doesn't appear to provide any call disconnect supervision. Therefore, if a nortel phone user calls a SIP phone as below
[nortel ph.]->[Norstar]->[ATA2]->[VOIP]->[sip ph.]
and gets forwarded to voicemail (the VOIP has an integrated voicemail box), there is no indication to the VOIP server that the caller has hung up. You end up with a message that is about 10 minutes long - 15 seconds of voice follwed by a click and a dialtone.

Feh seemed to have found a solution - anybody care to explain what he did?

Welcome to the board JordanP!

As feh mentioned in his last post, he configured the station to "receive tones". That is a parameter that is configured on a set-by-set basis in programming. It allows the default short tones to be sent through an ATA.

On a call-by-call basis, the long tones feature can be used on internal calls. External calls to the ATA will pass tones by default.

Your biggest problem will be the disconnect supervision. The ATAs were purpose built to be station level devices. You need an interface that can behave like the network. Try a T-1 or PRI.

Good Luck!

Welcome JordonP

I have successfully married VoIP to a Norstar using a 3rd party VMI box (trying remember the brand). The reason I didn't use Nortel VMI was because although it does provide a disconnnect signal, it does not use looop drop for disconnect, but you must tell the VMI which DTMF tone to send upon disconnect. The VoIP server I use could not use a DTMF disconnect signal.