sundance-communications.com/forum Forums TDM Business Telephone Systems NEC Telephones & Systems Asterisk to Nec DX2NA-24 via Sipura 3000

Hello,

I have a pair of Sipura 3000 ATA's connected to extensions 319 and 320 on the Nec PBX. They are registered to my Asterisk call router and appear to be working properly.

When I call the SPA-3000, I expect it to ring through to the Nec PBX, but the PBX never sees the ring.

The Nec PBX is effectively a black box to me, as I am not Nec trained. I assume that Ext 319 and 320 are FXS ports, so I have them cabled to the FXO ports on the SPA-3000s.

When I cable them to the FXS ports on the SPA-3000, they immediately go off-hook.

My goal is to have a call coming from my Asterisk PBX to the extension on the SPA-3000 appear as a call made from that extension.

Do any of you know what I'm missing?

Thanks.

Darren

Ok you have me confused. Your FXS ports have to goto a CO-Port or FXO if IP on the PBX. The ANALOG extesions on the PBX have to goto your FXO on the Asterisk. Does this make sense?

Also, just because you provide ring on an FXS, doesn't mean it will be telco grade ring current and will be detected by the CO card of the PBX.

I hope this helps.

Yeah, my earlier description lacks a little.

Here's a diagram:

Site 1 Site 2
====== ======
Asterisk <---Internet---> SPA-3000 -- Ext 319 on Nec PBX

Is 319 a FXO or FXS on a this Nec PBX? The SPA-3000 has 1 each of FXS and FXO ports, so I should be able to make it work no matter what type of port ext. 319 is.

Darren

An extension has to be analog on the PBX thus you need to provide an IP FXO. When you go off hook on the Asterisk you should pull dial tone from the NEC PBX. This is nothing more than an IP version of an OPX. Understand, the NEC MAY not trigger enough ring voltage to provide CO grade ring current to ring the FXO and THUS trigger where you have those trunk ports ringing on your Asterisk system. This is pretty easy...do you have a telco background?

Unfortunately for this specific case, I have a total IP background. So, the Asterisk side of things comes pretty easily, and simple PBX replacement is easy. But as to the details of how non T1/PRI telephony works, it's relatively new. In particular, NEC PBX's give me fits in that their totally closed systems.

What you are doing I have been doing for about 7 years or more. I have time on both sides of the fence and the logic of VOIP is relatively flawed in some aspects and justified in really weird ways. In a perfect world VOIP should work as designed. In the real world it's a different story and I think we have all had horror stories of the flaking T1 or router that is a version of IOS off.

Just remember FXS->FXO and FXO->FXS or analog to CO-port or CO-port to analog.

I've gotten a little further on this. I now have an indentical Sipura SPA-3000 in my office. By setting an extension on the Line 1 port and a different extension on the PSTN Line port, and turning one-stage dialing off on the PSTN Line port, I can dial the PSTN extension, it answers and gives me dialtone. If I dial a phone number, it drops it on my Verizon line.

At my customer site, I configured the same way. Dialing the PSTN Line port, which should be connected to a station port on the NEC PBX, I get no answer.

My 3000 at home shows line voltage of 0, where the one at the customer site shows -38.

From an earlier project, I had notes that told me to set the PSTN Line -> Line In-Use Voltage to a
number less than the shown voltage, -38 in this case.

The default is 30. It works with Verizon. Against the Nec PBX I have used 0, 30, -30, 48, -48. It still does not answer.

Thanks for your help.

Darren

First,are you using an analog port on the NEC PBX? Can you plug a a regular analog telephone into it and get dial tone?

> At my customer site, I configured the same
> way. Dialing the PSTN Line port, which should
> be connected to a station port on the NEC
> PBX, I get no answer.

So you have the PSTN port (FXO) port on the Sip3000 connected to an analog station port on the NEC 28i? Is that correct?

You can't really "call" the FXO port, it has no way to generate ring voltage, and the analog station port has no way to receive ring voltage. The FXO would have to set setup to go off hook and send a DTMF string to the NEC when it is "called" from the VOIP side. Is this what you're doing?

If you want to connect to the FXS port of the SIP3000 and be able to "ring" into the NEC system, you'll have to connect it to an analog line card (4ATRU) on the NEC. When it "rings", the call will follow the ring group routing setup in the NEC switch.

One gotcha I've found with trying to connect analog equipment to the analog station ports of an NEC/Nitsuko iSeries Switch is that the switch does not provide CPC (Calling Party Control) disconnects. With a proper POTS line, the CO should drop look current for a half second or so when the remote party disconnects. The iSeries doesn't do this and simply returns immediately to a dial tone after the disconnect. This can cause real problems for FAX machines, modems, answering machines, and other non-human equipment. I expect the VOIP gear might have a similar problem detecting the end of calls because of this.

I've tried everything I can think of to get a proper CPC, but haven't been able to. Anyone else had better luck?

Yes, that is correct, as far as I know. I do not factually know how to tell what kind of card I'm plugged in to. It was the last two free ports on the PBX.

I'm not sure I understand the mechanism, but I know what I want for results. I have an extension assigned to the PSTN port, and it's set to two stage dialing. I dial it's extension, it answers and gives me dial tone. That works connected to Verizon, but does not when connected to the NEC PBX. That would be an acceptable behavior.

I'm not married to any specific method. What I need is to get it so that location A can call location B via Location C. Location A is asterisk PBX at remote office on a dynamic IP address. Location B is the Nec PBX at the main office, where the SPA-3000 is. Location C is my Asterisk call-router on a static IP on the Internet.

I've run into this before. The SPA series has settings that can minimize the impact of this. My inexperienced guess is that it's a fairly common problem.

Thanks much. I really appreciate everyone taking an interest and helping me learn this and make it work.

Darren