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  #1 (permalink)  
Old March 30th, 2004, 01:02 PM
sipcpe sipcpe is offline
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sipcpe
Default FX-200 Trouble Shooting Tips

Sipura SPA 2000 does not recognize your PSTN DTMF tones on Transit In calls: When you place a Transit In call you should hear dial tone from your Sipura. When you press any key on your remote phone the Sipura's dial tone should turn off if the Sipura has recognized your PSTN DTMF tones. If this does not happen then you will need to increase the FXS Port Input Gain. To increasing the FXS Port Input Gain: Access

Last edited by eric : September 20th, 2006 at 01:09 AM.
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  #2 (permalink)  
Old May 7th, 2004, 11:05 PM
Anonymous Anonymous is offline
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whe I call through a third telephone line to the line 2 phone in the FXS port converter (12 volt), the phone rings and the DTA-310 from packet8 never gives me a dial tone, how can I fix this?
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  #3 (permalink)  
Old May 10th, 2004, 03:50 AM
sipcpe sipcpe is offline
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Default Helpful info to include with every support request post:

I am a bit confused and do not understand clearly what problem you are having. Please revert a detailed discription of your setup and problem you are having to support@sipcpe.com



1. What Gateway/ATA your are using.
2. What service provider you are using.
3. Where the unit is located ie what country
4. Confirm you have power to the units
5. Confirm you have installed and tuned as per product instructions.
6. Confirm the gateway and service is working properly without FX-200 installed.
7. Identify the problem as a Transit In (pstn to voip) or Transit Out (voip to pstn) call problem.
8. If a Transit In call problem are you calling from a cell phone or landline?
9. Discribe in detail what happens when you call into the FX-200. (rings x times, dead air, can't hear dial tone, no response to dtmf tones, etc.)
10. Send us your voip congtact number and best time to contact you and or remotely test the unit.
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  #4 (permalink)  
Old July 21st, 2004, 02:46 PM
UmerRazi UmerRazi is offline
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I am using FX-200 with Sipura-2000. In the begining it was not taking PSTN tone then I have changed FXS Port Input Gain to new value=3 instead of -3. Now it can take PSTN tone but when I call any number I receive the recording say "the # you have dialed is invalid" but when I dial the same number without going through FX-200 then I can connect to that # without any problem. Can anyone help me to find what is wrong with my setting. I have VOIP service from ICONNECTHERE company.
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  #5 (permalink)  
Old July 22nd, 2004, 05:59 AM
sipcpe sipcpe is offline
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Suggest you try to call the pstn number from a dif phone. Some cell phones and cordless phones have too much static on the line so the dtmf tones get garbled and a digit or two is lost or incorrect. If this does not work for you send an e-mail to support@sipcpe.com with your contact details and a tech will contact you to assist you.
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Old July 22nd, 2004, 05:59 AM
Linksys SPA3102-NA (Unlocked)
Includes VoIP/PSTN gateway, FXO/FXS ports, and router.
Sale Price: $76.95
  #6 (permalink)  
Old September 6th, 2004, 03:19 PM
syedamerali syedamerali is offline
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Default Low Volume

I am using a cable internet connection which is 100 kbps both up and down. The modem is connected to a Dlink 624 router to which is connected a Cisco ATA 186 which is running two seperate Stanaphone numbers on Line1 and Line 2. There is no port forwarding, neither is the ATA in a DMZ; that does not bother Stanaphone as I seem to make and receive calls without the slightest issue. (There are issues with Stanaphone which is another story).
The FXSInputLevel on the ATA is set at +2.
The FXSOutputLevel is set at -1
PSTN is connected to Line 2 of the FX-200
Line1 is connected to line 1 of the ATA
The FX200 has been tuned as per instructions - but a solid yellow S2 was unachievable - it is almost solid.
When using transit in from a cell phone, a hard wired phone or a cordless phone the dial tone from the ATA does get through the FX 200 but is very very faint. Further, DTMF gets through to the ATA very erratically no matter what phone one uses. When dialling direct from the ATA there is not the slightest issue.
On transit out the dialtone of the PSTN is broken and No DTMF gets through. Stanaphone specifies DTMF to be set to inband to which the ATA is set (AudioMode is set to 0x00000000).
I am in Pakistan and can be reached on sip:67166@fwd.pulver.com or 1 248 724 4483 or sip:6467229920@sip.stanaphone.com. Pakistan is on GMT+5; a good time to call is 9am-12 noon New York time.
Any help would be most welcome before I write off the FX-200 as a dead loss.

PS
To transit out on the FX-200 dial sip:6467229921@sip.stanaphone.com - on the dial tone dial 7589205 this should get you through to me.
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  #7 (permalink)  
Old September 7th, 2004, 03:04 PM
sipcpe sipcpe is offline
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Someone from support will contact you shortly on this. We have had issue with the Stanaphone service in the past, can you send me their url. Are they an open service? If so we will try to configure a box and do some testing with their service.
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  #8 (permalink)  
Old September 7th, 2004, 04:12 PM
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PhoneBoy PhoneBoy is offline
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They allow anyone to sign up and were providing about $2/mo of calling as long as you used the account every 30 days or so.
__________________
Technical questions should be posted to the forums, not sent via PM to me.
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  #9 (permalink)  
Old September 7th, 2004, 04:35 PM
syedamerali syedamerali is offline
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I got a call from support asking for the serial number of the FX-200 and my postal address in Pakistan so that they could send me a replacement ..... Pretty efficient if something comes of it.

The URL of the Stanaphone service is www.stanaphone.com
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  #10 (permalink)  
Old March 7th, 2005, 11:21 PM
markos007 markos007 is offline
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hola can not enter to the advanced features i dial ## and no answer is new and i never put it a password did he have a fabric password?
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Old March 7th, 2005, 11:21 PM
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