Hi folks.

I have a Nortel 6x16 KSU, and a NT8B90AC ATA plugged into one of the ports (with an analog phone on the other side of the ATA).

I've noticed that when a call is placed from a Nortel phone to the ATA, that DTMF tones are not passed through to the analog phone.

Is this a configuration issue? I want to replace the analog phone with a different device that needs to be able to receive DTMF info, and right now the ATA isn't allowing that.

Thanks!

What type of device do you want to connect to the ATA? Some type of Voicemail?

When you say "from a nortel phone to the ATA", do you mean internal intercom calls?

If you have an ATA with an analog cordless unit connected to it, and I call you , and you answer, you will hear DTMF on your end, if I start pressing digits on my end.

For troubleshooting purposes, I have a regular old analog phone connected to the ATA.

What you describe above is what I'd like to have happen, but is not. A call is placed from a phone to the ATA...the analog phone rings...I pick it up, and the call is in progress...I press digits on the phone that placed the call, but the DTMF tones are not heard on the analog phone.

Any idea why?

What Rich said. :thumb:

I'll just add: If you are trying to test your application from an internal phone, you can send tones through the ATA by invoking the 'Long Tones' feature (808).

Good Luck!

An ATA will not let the DTMF go through. But you can use a VMI (voice mail interface) unit that will let DTMF pass. It was designed for use with 3rd party voice mails and the Norstar.

Crud. I wanted to be able to use an ATA.

What is the difference between an ATA and a VMI?

Sure it will, by default from external calls and with 'Long Tones' from internal calls. Is your application internal or external?

The VMI will act just like an ATA to the analog device. But you can also go in with Feature codes and make changes to it so it will communicate with voice mail systems to do Message Waiting Lights and thngs like that.

It's an internal application.

So, is the "long tones" feature something that needs to be configured on the ATA, or the calling station?

Long Tones is only a call by call basics. I'm sorry I thought he was looking for somthing that allowed the tones all the time.

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.

BCM, How do I go about programming the ATA station to "receive tones". Or where can I find documentation to show me how to do this. I dont want to have to dial Feature-808 for every call.

nevermind - got it

I've seen this "Receive Tones" option before, but I was looking for it the other day (on new software - i think) and couldn't find it. Do you guys know what software levels provide this feature?

I don't have a 7.0 Norstar in front of me right now...but it is definately an option in BCM50.

anyone familiar with this? I suspect that not all norstar software versions support "receive tones" but I'm not sure.
Off topic a bit - what capabilities does the norstar have for forwarding incoming calls to an outgoing line as opposed to forwarding to an extension. I'm considering using a few free FXO ports on the Norstar to handle the dialing into my SIP server, but only if I can transfer incoming external calls to (as far as the norstar is concerned) another external line.

Trunk to trunk can be done by via external call forwarding of a digital set. Provided you are using sftware 4.1 or higher. Keep in mind that this will create a conference between the 2 FXO ports and you WILL have sound level issues.