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MagicJack and SPA3102 FXO

 
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xbzus
MagicJack Newbie


Joined: 06 Jan 2009
Posts: 1

PostPosted: Tue Jan 06, 2009 9:45 pm    Post subject: MagicJack and SPA3102 FXO

HI,

I fixed the MagicJack to a Linksys SPA3102 FXO. When I dial any number from my phone attached to PBX, I'm just getting the FXO's dialtone. While listening to this dialtone, I go to the PC where MJ is fixed and dial from the softphone, then it connects both the PBX phone and callers phone. When I hangup the PBX phone MJ too ends the call.

I believe that my PBX phone is correctly connecting to the SPA3102 and 3102 in turn connects to the MJ but MJ doesn't recognize the key presses from the PBX phone passed through the SPA3102. My SPA3102 shows the last called number correctly.

Is there any specific regional setting for MJ? or whats the problem

thanks
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bgarczynski
MagicJack Newbie


Joined: 25 Mar 2009
Posts: 5

PostPosted: Fri Mar 27, 2009 5:52 pm    Post subject:

I do not have a solution to this problem but I have the exact same issue. I have been told it’s an issue with the MJ sending only -24 volts rather than the normal -48 volts a regular POTS line sends. I don’t know if this is true as I have no way of changing the voltage my FXO port senses. For some reason DTMF is not accepted by the MJ from FXO ports. The issue is called “Unbreakable Dial Tone” but I have not yet been able to resolve it. Please post if you are able to find a fix or work-around for this issue. I will let you know if I find it as well this is driving me crazy!
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mberlant
Dan Should Pay Me


Joined: 01 Feb 2009
Posts: 829
Location: Japan

PostPosted: Fri Mar 27, 2009 7:40 pm    Post subject:

bgarczynski wrote:
I have been told it’s an issue with the MJ sending only -24 volts rather than the normal -48 volts a regular POTS line sends.
In a Sipura/Linksys device, like the one xbzus uses, it is a matter of changing "Line-In-Use-Voltage" at the bottom of the PSTN Line page from the factory default of 22 to the "works with 24v systems" setting of 11.

This is not necessary, though, for most users of an FXO device, because most people are trying to interface their MJ account with their POTS line, and the POTS line is always -48V. I can't think of any reason to physically connect an ATA's FXO port to an MJ dongle instead of just putting the MJ credentials into that ATA.

bgarczynski wrote:
For some reason DTMF is not accepted by the MJ from FXO ports.
Also on the PSTN Line page, about in the middle, are some DTMF processing settings. For inbound they should both be Yes, and DTMF Tx Method should be Auto, but you may need to force a particular method to make your particular configuration work.

p.s. Could someone please describe a scenario where it is preferable or necessary to physically connect the LINE jack of the SPA to the MJ dongle instead of putting the MJ credentials into the SPA?
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bgarczynski
MagicJack Newbie


Joined: 25 Mar 2009
Posts: 5

PostPosted: Sat Mar 28, 2009 10:36 am    Post subject:

It is preferable to connect an FXO port to the MJ in cases of lab. I am studying for the CCVP certification and want a cheap analog PSTN connection. Cisco FXO ports do not have the ability to change the line in use voltage. I can not interface directly with the MJ account from my Cisco ATA/Gateway with out creating a very complex configuration beyond the scope of my lab. I simply do not understand why the voltage for line in use matters in this case. Why is it so difficult to pass regular DTMF tones after the MJ goes off hook? Obviously the line works for incoming calls does the the line in use have no affect for this type of call? DTMF passes into remote IVR systems just fine through the MJ. Is there not some type of repeater that could be placed between the FXO and the MJ? It just doesnt make sense to me why an analog phone's DTMF is accepted by the MJ but the FXO DTMF is not.
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mberlant
Dan Should Pay Me


Joined: 01 Feb 2009
Posts: 829
Location: Japan

PostPosted: Sat Mar 28, 2009 8:17 pm    Post subject:

The Line-In-Use threshold is used for the FXO device to determine if another device bridged on the line is already occupying it. This may be an extension phone, alarm panel, modem, etc. If you are reliably picking up the MJ "line", then this is already probably set at a good value.

Next I would check for two values: DTMF gain and Interface impedance. If the gain is too high or too low the MJ dongle won't reliably decode the DTMF. Also, if the interface is wrong (900 Ohms, or one of the composite European values) the DTMF will be hurt.

How is the call clarity on an incoming call? Is it perfectly clear (the telephone interface part)? Or, does it sound just a little bit fuzzy? Fuzziness is not usually an artifact of digitization, but of impedance mismatch.
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bgarczynski
MagicJack Newbie


Joined: 25 Mar 2009
Posts: 5

PostPosted: Sun Mar 29, 2009 7:26 pm    Post subject:

I have not noticed any clarity issues through the FXO to MJ line during a call. I have tried numerous gain/impedance but have not found one the magic finds acceptable. Do you know what an optimal gain/impedance setting would be for use with the MJ?
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mberlant
Dan Should Pay Me


Joined: 01 Feb 2009
Posts: 829
Location: Japan

PostPosted: Sun Mar 29, 2009 8:54 pm    Post subject:

Impedance must be 600 Ohms. 0dB variance should be ok.

You mentioned that clarity seems fine, which is good for you but bad for troubleshooting.

Could you please confirm whether the FXO is sending DTMF that the MJ does not recognize or the MJ is sending Dial Tone that the FXO does not recognize?

Could you also please hook up the FXO to a real POTS line for just long enough to confirm that it is not the culprit?

Also, can you confirm or deny that incoming calls are detected and answered properly by the FXO?
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bgarczynski
MagicJack Newbie


Joined: 25 Mar 2009
Posts: 5

PostPosted: Mon Mar 30, 2009 8:14 pm    Post subject:

I dont have a regular POTS line to test. The FXO does detect incoming calls properly. I have proven this as the calls are PLAR'd to a pilot point and then routed like a normal call in my lab environment. I do not know how to test if the FXO is not detecting the MJ dial tone properly or not. My router thinks the call is up and its streaming RTP but in the RTP all thats played is the MJ dial tone.
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mberlant
Dan Should Pay Me


Joined: 01 Feb 2009
Posts: 829
Location: Japan

PostPosted: Tue Mar 31, 2009 12:40 am    Post subject:

Pardon the cynicism here, but perhaps this is part of the test. The MJ dongle only knows how to put out standard North American Dial Tone (350Hz+440Hz), so the next thing I would check is to make sure that the FXO's Call Progress Tone settings are set for the US. If the FXO were set to expect a Japanese Dial Tone (440Hz), for example, the absence of the 350Hz component would prevent detection.

You mention that there is an RTP stream that you can capture and observe to contain the MJ's Dial Tone. This comment raises my other eyebrow. Assuming that this RTP stream is inside the Cisco router under test, where is that RTP stream going to? During call setup there is no RTP stream yet established. This comment indicates that you may have an erroneous hotline defined within the Cisco. If there is such a hotline, the FXO will connect the calling party to the MJ and establish the talk path without any Dial Tone/DTMF interaction. This is normal operation for that abnormal configuration.

The normal call establishment procedure is for the originating ATA to dial a call and for the Cisco router to be programmed to translate the digits for outpulsing onto the CO trunk. For example, the originating ATA dials 8-1-212-555-2368. The Cisco uses the 8 to route the call to the first available MJ FXO port, the FXO port listens for Dial Tone, and then the FXO port outpulses the 1-212-555-2368 onto that trunk. The FXO port then listens for Call Progress Tones on the line prior to establishing a talk path. If it hears Busy or Fast Busy it will send an appropriate information packet back to the originating ATA and that ATA will play the Busy Signal or Fast Busy Signal into the caller's ear.

Only once a connection is reasonably expected will the Cisco network establish the talk path. An abnormal configuration would be to set up a hotline path between the originating ATA and the terminating FXO port and for the human to hear the distant CO Dial Tone and further dial the destination phone number. If this is your scenario you must figure out which of the three DTMF encoding schemes is compatible with your chosen CODEC and make sure that your ATA and your FXO port are both configured to utilize this same encoding scheme.
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