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Linksys SPA-3102_unable to configure

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MikeMelga

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Hi.

I am using 3CX for some time now. I have a analog gateway (SPA-3102), a GSM gateway (Portech MV372) and 2 voip lines. Everything is working ok but one of the things I still haven't been able to do is properly configure the SPA-3102.

Currently I am able to both receive and make calls in 3CX using the SPA-3102 with no problem.

My problem is that I have an analog phone connected to the SPA-3102 that I never got it to work.

Can I have that analog phone configured as an extension? What I would like to do is be able to dial and receive calls on that phone.

How can I do that?

I had some help some time ago here:
http://www.3cx.com/forums/integrating-ata-gateway-and-2-ata-phones-33055.html
but never got it work :(

Can someone help?

Thanks.
 
I think in the setup they mention that with 3cx and the spa32102 you should use it as ata device or a gateway but not both. kind of hard to believe since that's what the device was designed to do


Note: The Linksys SPA3102 can either be set as a gateway OR as an ATA device. 3CX does not support the device when configured in both modes simultaneously.

http:/ /www.3cx.com/voip-gateways/linksys-spa3102/ - NO LONGER AVAILABLE



jasit
 
Jasit,

Thanks for the reply.

I know it does say that but I've seen several testimonies that it does work with no problems. Unfortunately, none explains exactly how to achieve it :(
 
I've got several with both functions working. One key is to use the correct port numbers for each "device". There are also a couple of default settings that need to change to isolate the ATA from the gateway. The internal dialplan must also be "crafted" as to prevent calls from routing directly out on the FXO.

You will need to provide more information as to what errors, or conditions, you are getting when trying to place calls (or even register) with the FXS portion.
 
leejor,

Thanks! Nice to ear from you again.
Tomorrow I will post here my current settings and what my problems are, exactly.

If you can help...I'll be most appreciated.

Thanks.
 
Ok.

So....I've made some tests just to remind me of what was wrong.

When I pickup the handset I have no dial tone.
Also the headseat does not ring on incoming calls.

When I first configured the gateway I've followed these steps:

http: //www.3cx.com/voip-gateways/linksys-spa3102/ - NO LONGER AVAILABLE


and it's working since... but with absolutely no function on the analog phone connected to it.

Can you post your settings?


Regards,
Mike
 
No dialtone would indicate (for starters) that registration is failing, for some reason.

Is there an indication of a registration attempt in the 3Cx logs? Something in there, if there is anything, may give some clues as to what's happening.

Obviously that server name (Proxy field), or IP, will be that same as in the FXO settings, so the extension number (User ID), and Password are going to be unique for this extension. Of course, up at the top of that page, the Line must be enabled.

In the PSTN tab, under PSTN to VoIP gateway Setup, be sure that PSTN ring thru Line 1 is set to NO.

If you still find you aren't getting anywhere, post the Line tab settings of your 3102.

The settings in the 3CX link you posted should work with no problems for the ATA settings. the fact that you have the Gateway functioning properly is a good start, it is normally the biggest obstacle most users of the 3102 come up against. The ATA part is relatively simple in comparison.


I should add...as it came up as the cause of problems a while ago...you do have the Ethernet cord plugged into the Blue socket, correct?
 
leejor,

Yes, I have the gateway connected on the blue socket. However I only have it connected on the blue socket. The yellow ethernet connector is not connected. Seemed a bit strange right from the beginning why a device should have both an ethernet and internet connector but...

Anyway... last night I did some more testing and came to other conclusions: I do have a dialing tone on the 3102. I connected an old analog phone and it did have a dial tone and I managed to make a call. So... it seems the problem is with my wireless phone. It is a Bang & Olufsen Beocom 2 which works fine if I connect it directly to a regular phone port.

I've noticed the dial tone of the 3102 is somewhat 'strange'. It is not exactly like the landline one. I'll try to see if the 3102 has some setting regarding the dial tone.

One other thing: when I made a call from the analog phone it reacted just like a regular phone without 3CX. Is there any way I can make a call from the phone using 3CX outbound rules?

Thanks!

Regards,
Mike

edit: Ahh....and although I have a dial tone, the phone still does not ring on incoming calls.
 
Ok...status update on this.

I took some time today to get this working.
I now have a fully functional analog line using the old analog phone. It rings on incoming calls and is able to make calls using 3cx dial plans.

However....using the wireless phone I can only receive calls and without any caller id (I guess that also happens on the old analog phone but it doesn't have even a display). When I try to make a call... I enter the numbers and when I press dial I almost immediately ear a busy tone.

I'll continue my experiments. Any help is always welcome :)

Thanks.

Regards,
Mike
 
Ok...got it.

Changed the dialplan from [x*]. to [x*][x*]. to allow more time for the wireless set to dial. Fixed dialing out.
Played with the callerid format until I got one that worked. Solved.

Everything works now :) Well....almost everything. I am still strugling with bad disconnection detection when the calls originates from the PSTN but that is already an old problem. Will deal with it in time.

Regards,
Mike
 
You caller ID problem may be the Regional setting or a timing issue, depending on how CID is implemented by your provider. There has to be enough of a delay, to capture the CID, before 3CX is notified of the call.

As far as your dialling goes...the dialplan suggested by 3CX, is a very basic "catch all" plan. You may want to customize your dialplan to more suit the numbers you dial. That way, you don't not have to wait a few seconds for a number to be sent, or hit the octothorp (#) at the end, to send the call on quickly. There are a number of sites with good information on this. These are just a few.

http://www.solidfluid.co.uk/sfsite.php/00000223

http://www.cisco.com/en/US/products/ps10033/products_qanda_item09186a0080a35a44.shtml

http://www.netphonedirectory.com/pap2_dialplan.htm

While dialplans work pretty much the same way in Sipura/Linksys/Cisco ATA's and VoIP phones, the 3102 adds a few extra commands. It has the ability to send calls out on more than one VoIP provider, and to send calls out directly on the FXO port (PSTN line), because of this , there are some added commands applicable only to the 3102.
 
leejor,

Thanks for all your help.
I guess I have the hardware part solved. Now it's time do dig in other options :)
 
MikeMelga said:
Yes, I have the gateway connected on the blue socket. However I only have it connected on the blue socket. The yellow ethernet connector is not connected. Seemed a bit strange right from the beginning why a device should have both an ethernet and internet connector but...

Sorry, I should have explained this a bit.

The 3102 along with some other Cisco devices have a built in router. It allows the Blue port to be connected to a modem to pick up a public IP resulting in no NAT for VoIP calls. The yellow socket can then go onto a switch to "feed" other devices on your network providing DHCP as with any other router. For most users that already have a router, this is not needed. There is an option to change the yellow socket to Bridge mode, which is what I do. This allows you to add a second device that simply passes though the 3102. This is handy if the SIP device is located under a desk and there is only one Ethernet connection. The computer on the desk can connect, over the same network via the yellow socket.

It also allows you to daisy-chain devices if there is a shortage of Ethernet ports.

Of course before you change the port to Bridge mode, you must enable access to the device from the WAN (blue) port.
 
The SPA-3102 is a pain do configure.

I guess I will try to mess with it as little as possible. I'll use the 'If it works... don't touch it!' technique.

I have turned myself to the CDR output section. I am trying (without success) to make a socket connection to my application. Still haven't managed to get any output. Do you have any experience with the CDR output?

Regards,
Mike
 
Sorry, no need (up to this point) to delve into that area, so someone else will have to step forward. You might want to consider starting a new post, with a new subject line, to attract the help of someone familiar with CDR.
 
SPA-3102 is extremely easy to configure, you need to configure it manually. Read Cisco / Linksys admin guide instead.
There are 7 different modes of operation and is absolutely possible to use FXO & FXS ports independently.

I have over a 100 of them installed into past few years and none of them had any problems.
This topic has been discussed many times into forums.
 
eagle2,

I have my 3102 working ALMOST as it should.

My only problem is with the disconnect tone handling. The 3102 fails to detect the disconnect tone. When I a calls comes from the 3102 line it starts ringing. If I pickup the call and hang it myself...no problem. If the calling party disconnects the call...it keeps ringing and eventually gets picked up by the voice mail, which adds more minutes to the call.

The disconnect is set to the recommendations for my country (as the manual refers). I've also tried many more tones which all should work but they don't.
I've read that it could be a firmware problem:
https://supportforums.cisco.com/thread/2028099
so I downgraded FW up to version 3 with no success.

I am starting to give up on the SPA-3102. Don't know what else to do.

Do you have any ideas?

Thanks.
 
Short of a faulty unit, disconnect tone should work as long as it is set correctly. the tone can vary from provider to provider in the same country, and even from central office to central office depending on the make of switch providing dialtone and the competency of the company that installed/maintains them.

So...be sure that your 3102 is set to work with what is being sent.

Short of the disconnect tone, the next step is to make use of silence detection. this is timed and options usually allow monitoring of the VoIP or PSTN direction. The drawback is that it can drop a call, if there is a long lull in the conversation. I've found that being on hold,listening to very low level music, can also trigger a disconnect.

The best method is CPC, which puts a short open on the line when the remote party hangs up and you don't. Unfortunately, all providers don't offer that option.
 
Disconnect tones may vary from country to country and from provider to provider. The optimum solution is, if predefined settings not working, is to record the disconnect tone (can be done with 3CX) and to analyze the recorded sound. It is relatively easy to estimate frequency or frequencies and duration of tone / pause with some more advanced sound editor and to put the necessary settings into SPA-3102 (usually 2 different tones like 440 Hz and 620 Hz, 0.5 sec on, 0.5 sec off). Depends too much on provider.
 
leejor and eagle2,

Thanks for the reply.

So...I will start making sure I am using the right disconnect tone. How can I record it and which sound editor can I use to analyze it?
 
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