[syscon]

Originally Posted by PhoneBoy

We've talked about doing a configuration guide for the SPA3000 and Asterisk. It's something on the "to do" list.

That is exactly what is needed and not an ordinary configuration guide. It needs to be Gentoo type Handbook manual, so everybody can understand.

I've noticed that when I go through your webpage SPA3000 php setup script my Primary DNS and Secondary DNS are that of my ISP provider eth0 on my firewall, if I connect the device to my second network switch (I have two providers Cable and DSL) the Primary DNS is my gateway eth1
These setting will definitely have an effect on the configuration setup and need to be taken under consideration.

#Joseph

[bernart]

When FXS is configured to communicate with *box I can't get the SPA to apply the dial plan and send certain sequences out the gw0. Why is this? What's strange is that the sequences intended for gw0 is not being sent to *box either. So where is it going and why?

this is my dial plan
(xx.|[2-79]11<:@gw0>)

I want the 211 ... 911 sequence to go out gw0 over the PSTN line and the remaining sequences to go to *

The PSTN line is connected to the SPA

[mberlant]

Bernart's dial plan brings up a good question about Sipura dial plans. Is there precedence in a Sipura dial plan? That is, are the dial plan elements read from left to right, right to left, or in no particular order?

Is (xx.|[2-79]11<:@gw0>) the same as ([2-79]11<:@gw0>|xx.), or would this latter example solve Bernart's problem by evaluating the x11 case first and diverting those calls to the PSTN line?

[bernart]

Well it seems order matters in the dial plan. Flipped the sequence so that it looked like
([2-79]11<:@gw0>|xx.) and it worked. So it must be parsing the dial plan from left to right

[benjk]

Originally Posted by muppetmaster

Posted this on another site, though it might be useful here if not already embedded somewhere in a thread:

Sipura 3000 -> PSTN Config for Incoming

- Add a dialplan (ie - Dial Plan 1/Dial Plan 2) with a SIP URL destination as follows:

S0<:1000@192.168.1.50> (where the IP address is that of your * instance, also ensure the extension is properly accessible in you extensions.conf)

- Enable both the VoIP to PSTN Gateway and the PSTN to VoIP Gateway

- Set the default dialplan under PSTN to VoIP Gateway to the dialplan where you inserted the above SIP URL

Asterisk Configuration for Outgoing

- extensions.conf

[globals]
PSTN_GW=192.168.1.51:5062

[pstn]
exten => s,1,Dial(SIP/${EXTEN}@${PSTN_GW})

If behind a firewall this should be okay. If not, then of course you would want to configure with HTTP digest in mind.

This doesn't work for me, not for incoming calls anyway.

When I call in on the PSTN line connected to the SPA's FXO port, the SPA picks up the call but then immediately gives me a fast busy tone.

It doesn't even make any attempt to send the call anywhere.

On the "Info" tab under PSTN Line status it says ...

Last PSTN Disconnect Reason: PSTN Disconnect Tone

which leads me to believe that the SPA3000 cannot be used in Japan for incoming PSTN calls. It would seem to think that the caller has hung up, which is not the case. I can make outbound calls though.

Sipura support is not very responsive.

Has anybody else outside of the US experienced similar problems?

any hints appreciated
rgds
benjk

[benjk]

Originally Posted by PhoneBoy

We've talked about doing a configuration guide for the SPA3000 and Asterisk. It's something on the "to do" list.

Well, I was lucky enough to be able to borrow an SPA3000 which miraculously found its way to Japan and if I can make it work on a Japanese PSTN line, then we'll create a Sipura-3000 Assistant for Asterisk on MacOSX (with an outlook to port to OpenStep). A setup guide would be a byproduct of that work.

But if Sipura continue to be as unhelpful has they have been, then I may abandon this project. Right now it looks as if the SPA cannot be used on Japanese PSTN lines. If so, then I'd have no way of testing.

I think it would be a good idea to compile a table of countries in which the SPA can be used, with data from actual user reports. This should be along the lines of ...

Country
Telco (if multiple)
Type Approval (y/n)
Can dial out (y/n)
Can dial in (y/n)
Can detect CID (y/n)
Hangup Detect (stable/unstable/occasional or frequent false hangups)
Special settings (required FXO port parameters)
Comments (ie does not work well together with XYZ service option)

Once this information is available, I am sure there will be many more people who are able to contribute to such things as configuration guides for interoperability with Asterisk. The most important thing is to make sure that people know they can actually use the SPA before they order.

Sipura definitely didn't do their homework.

rgds
benjk

[mberlant]

What kind of support are you asking of Sipura that they are not giving? Are you asking for Japanese Caller ID support (which is not listed as an available feature)? Are you having trouble with Japanese call progress tones? Are you having trouble with impedance matching? Are you asking for Sipura to apply for Japanese type approval on a product that cannot be used with any licensed Japanese VoIP service provider?

The beauty of Sipura's product line is that it is flexible enough to be configured for situations that the manufacturer cannot imagine. The beauty of forums like this is that we all build on each other's experience and knowledge.

You accuse Sipura of not doing their homework. I suggest that this homework is yours to do. You started by privately importing a 3000 to Japan. You may very well have the only 3000 in country. I know that I don't have any in country. If you have specific interface questions, please ask them. Most North American PSTN settings work quite well in Japan. How are your problems manifesting themselves?

[benjk]

>What kind of support are you asking of Sipura that they are not giving?

Very simple. I am asking them to tell me whether or not the SPA3000 can be used to receive an incoming call when its FXO port is connected to a Japanese PSTN line.

As it stands, the SPA instantly hangs up on incoming calls and Sipura support seem unable to tell whether or not it is supposed to take calls from a Japanese PSTN line. My customer needs to know because if they wait too long, they won't be able to return the SPA and get a refund.


>Are you asking for Japanese Caller ID support

No.

>Are you having trouble with Japanese call progress tones?

I have used the wizard on Voxilla's web site to adjust the tones but the problem persists, so I guess it's not related to the tones.

>Are you having trouble with impedance matching?

Don't know. I tried all the possibilities (some 16 or so) and the one called "Global" works for outgoing calls. Yet, incoming calls don't work with that either.

>Are you asking for Sipura to apply for Japanese type approval on a product that cannot be used with any licensed Japanese VoIP service provider?

You will have noticed that this post is in a section called "Sipura and Asterisk". From that it should be obvious that my customer intends to use the SPA with his own Asterisk PBX, not with any Japanese VoIP service.


>You accuse Sipura of not doing their homework. I suggest that this homework is yours to do.

Maybe you want to read more carefully. The reason why I say Sipura didn't do their homework was clearly in the contect of not having any information of which features can be used in which countries. Basic features like "can make outgoing call" and "can receive incoming call" should be known to the manufacturer for the most important countries and this information should be made available to prospective buyers.

My customer bought this unit in good faith. He was talking to the Sipura reseller beforehand and the reseller was made aware that the unit would be used in Japan. Yet, they did not tell them that you cannot receive incoming calls on the FXO port.


>You started by privately importing a 3000 to Japan.

No, I didn't.

My customer asked Sipura for a local reseller and they referred him to two or three resellers in the US. They knew that he was calling from Japan.

>If you have specific interface questions, please ask them.

The question is whether or not the unit can *in principle* receive incoming calls when connected to a Japanese analog line.

If it cannot, I will tell my customer to send it back and ask for a refund.

If it is supposed to work, then I would like to know what settings need to be adjusted to be able to receive incoming calls.

The Info tab shows "PSTN disconnect tone" as a reason for the last disconnect. When I disabled detection of the disconnect tone, it showed "polarity reversal" as a reason for the last disconnect. When I disabled polarity reversal detection as well, it showed "VoIP call failed" as a reason. However, using a packet sniffer, I verified that the SPA did not make any attempt to send the call anywhere because there was no SIP traffic coming out of it.

It is registered with the Asterisk box, so the registration status would not have been responsible for it not trying to send an INVITE message.

To me, this all seems to indicate that the SPA has trouble with the electrical characteristics of the PSTN line in respect of incoming calls. For some reason it seems to get confused, it seems to detect a hangup condition that isn't really there and in turn it hangs up immediately.

Perhaps, this can be resolved by changing the settings under "International control". If so, I would expect Sipura support to be able to tell what settings to try. If not, I would expect Sipura support to be able to say so.

Instead, they take ages to respond and when they respond their suggestions do not seem to be related to the problem described. It's like they don't really know what to do and what to tell us.

rgds
benjk

[mberlant]

benjk, thank you for adding so many details in your last post. It's not surprising that Sipura has no answers for you, since it's pretty obvious that they have not yet begun investigating type acceptance in Japan. I don't know of any manufacturer who can tell the status of their product in countries they have not yet investigated.

Look on the label of the SPA. If you don't see a logo that looks like a Japanese Postal T (〒) at the right edge of a circle, the unit does not have type acceptance and Sipura simply does not have any answers. Japan may be an important country to you - it certainly is to me - but it is also one of the most difficult places in the world to obtain type approval for telecommunications equipment. When you consider the difficulty in getting type acceptance coupled with the lack of any sizable market in Japan due to Sipura not being included in the domestic VoIP market, it is not surprising that Sipura has put Japanese type acceptance farther down their list of things to do.

Even so, there is probably something else plaguing you. I have been connecting North American telephone equipment to Japanese lines for more than 25 years. 25 years ago there was an impedance problem that manifested itself in slightly fuzzy voice, but it did not even affect modem connections. I have encountered no problems with basic interconnection (making and receiving phone calls) over the last 10 years or more.

Have you looked into adjusting the "Line-in-Use" parameter? How about "Tip/Ring Voltage Adjust"? "Operational Loop Current Min"?

How about the Line 1 side of the SPA? When you plug a Japanese analog phone in and program the SPA for VoIP service on Line 1 can you make and receive calls? Is the voice quality clear? Does the ringer ring when it should and stay quiet when it should?

Did anybody test the unit in the US before shipping it out to you? There is a small possibility that it is defective. Do you have a PBX analog port available to you? Can you connect there and try the SPA?

I hope I gave you enough things to try. Remember, you are charting new ground in Japan. Without type acceptance Sipura can't legally give you any advice on how to make their product work in Japan, at least not if they ever hope to obtain type acceptance in the future.

Good luck.

Michael

[benjk]

>It's not surprising that Sipura has no answers for you, since it's pretty obvious that they have not yet begun investigating type acceptance in Japan.

I know from discussions with relevant people at Sipura that they have a pretty firm date for when they will have JATE type approval. I can't reveal any details though, as that would be up to Sipura themselves.

>I don't know of any manufacturer who can tell the status of their product in countries they have not yet investigated.

I have to disagree there. Any respectable manufacturer should be able to take a look at specifications of standards in a given country and be able to tell whether or not ***IN PRINCIPLE*** (!!!very very big emphasis!!!) their product is capable to fulfil those specs.

>When you consider the difficulty in getting type acceptance coupled with the lack of any sizable market in Japan due to Sipura not being included in the domestic VoIP market, it is not surprising that Sipura has put Japanese type acceptance farther down their list of things to do.

You are making a lot of assumptions there, most of which I would have to say are irrelevant and some of which are wrong. Anyway ...

I have been dealing with other manufacturers and the utmost that can happen is that they don't have access to Japanese standard specifications and ask me if I can get hold of it (in English) for them, which I can. Then when I sent them the PDF, they look at it and either tell me "should work" or "should work with changes" or "cannot and will not work".


>I have encountered no problems with basic interconnection (making and receiving phone calls) over the last 10 years or more.

I have encountered problems with false hangups before. In most cases manufacturers can tell you what to adjust to at least reduce those to a somewhat acceptable level.

>Have you looked into adjusting the "Line-in-Use" parameter? How about "Tip/Ring Voltage Adjust"? "Operational Loop Current Min"?

I have certainly seen those and I was thinking myself that those are the ones to adjust, but since I am not an electrician and I do not wish to damage the unit, I would expect Sipura to tell me what values to try.

Especially the ones with voltage would seem to carry the danger to lead to damage if set too low.

>How about the Line 1 side of the SPA? When you plug a Japanese analog phone in and program the SPA for VoIP service on Line 1 can you make and receive calls? Is the voice quality clear? Does the ringer ring when it should and stay quiet when it should?

FXS port works. Outgoing calls on the FXS port also work. Only incoming calls on the FXO cause imediate false hangups.

>Did anybody test the unit in the US before shipping it out to you?

Nobody shipped this to me. It's a customer's unit. They ordered it from a reseller in the US. We don't know if it was tested.

>Do you have a PBX analog port available to you? Can you connect there and try the SPA?

Japanese analog port -- same problem.

>Without type acceptance Sipura can't legally give you any advice on how to make their product work in Japan, at least not if they ever hope to obtain type acceptance in the future.

That's not true. I have all the documents from JATE on the approvals process (in Japanese, which is binding, while the English translations are not) and it says no such thing. I have also been talking to people at JATE.

It is perfectly OK for them to give advice how to configure a device to match Japanese standards.

Besides, if they were under the illusion that they couldn't, then it would certainly have been wrong of them to refer my customer to a US reseller to purchase their product. They should have said something like "you must not order any device and use it in Japan". They didn't do that.

rgds
benjk