configure SPA3102 for payment terminals

fdiogo

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#1
Hello all!

I have a problem with the configuration of Elastix+SPA3102

I have Elastix with a BRI card from Openvox B200p and a ATA Linksys SPA3102.

I have in the fxs port a payment terminal (don't know the name for this, is the machine that you use in the shops to pay with credit card, mastercard, etc.) but is not working :-(.

I can make calls from this line with the ata without any problem!

i can see in CLI that is calling a number for data ( like the old times when we connected to internet by pstn line ) but i always get a error on the machine. i have configure a outbound route so it always use the BRI line instead the voip trunk!! :)

What is the best codec for this?

Is any configuration in the SPA3102 that i need to change to get this to work?

is it possible to work??? :unsure:

Any ideas will be grateful!

Thanks for your time!!!!

Diogo
 

Patrick_elx

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#2
I don't have a spa3102 then I'm just throwing my two cents:

is there any fax detection on the spa or the extension that coulp mess or help with the modem?

Analog fax is not working really well over VoiP, modem should'nt be different.
 

dicko

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#3
How is the 3102 handling dtmf signaling, it would be safer to set it for inband for credit card machines etc. to work

also I believe that you can change the length of the dtmf tone, it should be at least 100 ms for many older systems.
 

fdiogo

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#4
Hello again!!!!

thanks for you help!!!!

I can send fax from this line with no problem even by a Sip trunk (Voipbuster)!!! but for the Credit Card payment terminal it will always go by ISDN!!

Patrick_elx -> the ATA SPA3102 has Fax detection and i can send with no problem!

dicko -> i have changed DTMF to Inband and the DTMF Playback Length to ".100". THERE WAS A PROGRESS:) !!!! Now in 10 attempts ONLY ONE WORKS! :(

we are getting closer!!!!! AND YES WE CAN!!!! :)

Diogo
 

dicko

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#5
Extend tone tp to 200 ms. and also check your codec order, I don't know where you are and your profile doesn't show but make sure you only hva ethe right g711 codec in your codec list, basically g711u for US and a few other places but g711a for most of the rest of the world.

Quick check would be to yourself on a regular phone and listen to what you hear,
 

fdiogo

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#6
Hi dicko !!!

I am from Portugal, Europe! i will try that tomorrow. in the ATA i'm using codec G711a, is this right?

in the phone i hear the normal tone, apparently. what am I supposed to hear? the normal tone or something else?

thanks!

Diogo
 

dicko

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#7
correct g711a for portugal

yes you should hear solid undistorted, unwavering tones , if they are long enough and of the right frequency there is no reason that the "terminal" you are talking too should not recognize them, I have occasionally found that you need a small delay before sending them to let the far end equipment to let it settle down, once again experiment with manually dialing the far end and notice any aberrant behaviour.
 

Chilling_Silence

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#8
Also things such as bad jitter / latency will have quite an adverse effect on those things. Play around with the jitter settings, as well as QoS :)

Also, using anything like compressed g729 probably wont work, stick with g711. Higher quality VoIP Providers also go a long way towards fixing those kind of issues, freebies usually arent the most reliable.
 

dicko

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#9
Thanks Chilling, all good points,

Perhaps to clarify this particular scenario, I believe ISDN trunks are natively (and exclusively) g711(a or u(properly the greek letter mu)) if you don't use this codec specifically from your endpoint, asterisk will have to transcode to g711(whatever), the flavor of g711 is locale specific, so from portugal ensure that the trunk has deny all and allow ONLY g711a, your ISDN provider will transcode, if necessary, if you call internationally. (also set the DTMF mode in the trunk to inband, (just for shits and giggles, I believe that the ISDN D(signalling) channel is only used for call setup/pulldown, so this point should be moot)

Jitter and QOS are a non issue in your scenario, i.e. BUS connections to Telco handoff (the hardware card) and LAN connection to the ATA (providing a relatively quiet LAN.)

But there are some linksys specific "feature codes" in the 3102 that should be eliminated from the device to ensure transparent flow of signaling over SIP to your box, and some settings that will ensure transparent transmission of DTMF over the connection to the far end terminal, forcing DTMF inband ensures that the tones you dial from your device are sent within the rtp load, not the SIP connection per se., this should be transparent, but I suspect that this hardware might intercept and modify these tones. forcing the tonelength above the default will help the signaling to be more robust, legacy equipment often needs tones as long as 150ms.

I specifically refer you to:-

http://www.voip-info.org/wiki/view/Asterisk+DTMF


That's my 2cents worth, good luck.
 

Chilling_Silence

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#10
If its a LAN-only connection to an asterisk box then to PRI / BRI, then I would disable the jitter correction entirely in the SPA3102 -- Its enabled by default and will likely be messing with things if thats the case.
 

fdiogo

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#11
Hi again!!

first want to thank all your opinions and suggestions! has been very positive!! :)


I have try your ideas but still no luck, i think it only works the first time.:-( i'm not sure....

in SPA3102 ther is a option "DTMF Playback Level:" .... what is this??? does it matter for this case?

dicko -> i have disable "Jitter Buffer Adjustment:" but i have "Network Jitter Level:" thant only have these options "low,medium,high,very high, extremely high" what is the best choice?

dicko -> SPA3102 has a lot o codes.... "Vertical Service Activation Codes" and "Outbound Call Codec Selection Codes" in each of this options there is more or less 25 codes, do you mean clear all?

i have call a techinal of the payment machine, and he tould me that the frequency was low.... the big question were i can change that....:blink: he told me to call my electrician to solve! eheheh!:)

I will continue trying..... i will keep you informed of my progress... :)

Diogo
 

Chilling_Silence

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#12
The DTMF playback level is how loud the pushes of buttons etc should be.
The Jitter Level should be set to "Low", there should even be an option to turn it right off.
 

dicko

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#13
I concur with chilling, disable the jitter buffer,

all vertical service Activation codes are probably unnecessary and might collide with Elastix's concept of what a dialed string will do, the codec selection codes are for overriding the default codec however almost certainly neither are relevant here. ( personally I delete them all)

DTMF playback level however could well be pertinent especially as your advice from the technician was that "your frequency was low", (his name wasn't Kenneth by any chance was it?) it would only be relevant to in-band signaling, which is what I believe you have had partial success with.

The technician is incorrect in saying that the frequency was low, as DTMF (Dual Tone Multi Frequency) is a standard used world wide since the 1960's and as these are hard coded into asterisk and have worked for everyone since, I can only surmise that he is spinning you a line, and provided that your computer hardware clock is not completely insane I suggest you discount his advice.

As a test I suggest you call some other DTMF enabled service, like a "bank on line" or any IVR based system like an airline booking service, if it works with them then you have gained some knowledge that your system is actually producing and sending across the ISDN connection valid codes.

Then you could try experimenting with the DTMF playback level.

However, in my experience, once you get the DTMF audio tones out to the tdm (ZAP/ISDN) trunk nice and clean and long enough, you're good to go. I have never had to adjust that setting, but there is always a first time.

It is perhaps possible that your payment terminal is mismatched with the ATA as to some arcane parameter , (you don't have anything sharing that line do you, actually if you cared to connect any old analog phone in parallel, you could listen to the real transaction between your terminal and the far end by picking it up (after the light changes color) and listening) and sipura who marketed the device originally have a lot of documentation "out there" on configuring the fxs port on this hardware (I think that some of that legacy documentation would be found with a search on SPA2000 (the granddaddy of all current sipura/linksys/(and even cisco now) ATA fxs devices))


(preemptively, and for anyone who is confused by my reference to "Kenneth", http://ask.yahoo.com/20010619.html)
 

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