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SPA3102 Doesnt see/pick up PSTN lineTechnical support, how-to guides, troubleshooting, and general assistance for Linksys hardware. |
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| Hey guys, Ive recently purchased an spa3102 and im having some fun with it Anyway, im trying to get something really basic working which is for when someone calls the analogue PSTN line, for the spa it to pick it up and forward it to in this case an asterisk box. I am not worried about the asterisk config or even the dial plans on the spa since the worrying thing is, the spa doesnt even appear to recognise the pstn is ringing, no lights flash etc, if i refresh the info page while im getting ringback tone, it shows on hook, 0V. if i plug my analogue phone into the PSTN line instead of the SPA its working fine (it rings etc), and i can make outgoing calls, so im sure its not an issue with my line. any ideas? or is my box fried maybe?? Is there a simpler testi can do, to completely forget the voip aspect just to see whether my spa is broken? Cheers, Chris ps im in the UK. |
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| Have you tried the Linksys SPA3102/SPA3000 Configuration Wizard for Asterisk ? It can provide you with a basic SPA3102 and Asterisk configuration to work from.
__________________ Eric Chamberlain, CISSP Chief Technical Officer Voxilla |
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| Hi Eric, Thanks for your post, yes i had a look at it but i am reluctant to use it for a couple reasons, firstly it wants voip account details, which i wont be using on the SPA (i use * to deal with that), and also im wary to simply blindly put in settings when i dont know what they are doing. anyway, after some reading around i have tried to make a simpler test, using an analogue phone on line 1, to see if i can get it to dial out. i added this entry to the dial plan (01xxxxxxxxx<:@gw0>) which im hoping will forward to the call to the analogue pstn when i dial 01 + 9 digits. When i do this i get fast busy, and the call state (on line 1) goes to invalid, and the pstn line stays as is, idle/0V etc i am really concerned my spa is completely unable to do anything with the analogue pstn which is basically the only reason i bought it! |
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2. The following settings are necessary on the pstn tab for you to be able to dial out using your dial plan. Line Enable YES Preferred Codec: G711u (or other if Use Pref Codec Only is NO) Dial Plan 1: (xx.) (or another valid plan) VoIP-To-PSTN Gateway Enable:YES One Stage Dialing YES Voip Caller Default DP; 1 (or some other number, not NONE) 3. You need an element in your Line 1 dial plan such as the one you indicated. The dial plan priority is left to right. In your case you should not have something to the left of the element you indicated that would satisy your pstn dialing and would therefore send it to the voip line. The Info tab, which you mentioned, should show a PSTN line voltage of about 48v on-hook, about 7v off-hook for a U.S. pstn line. A reading of 0v generally indicates that the pstn line is not attached to the "Line" plug or there is something amiss with your cabling. In order to hear ringing on an incoming pstn call, you need to have an analog phone attached and you need to have PSTN Ring Thru to Line 1 YES on the PSTN tab, and you need to have PSTN Answer Delay set to some reasonable number of seconds (16 is the default I believe). |
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| OK thanks for those settings ive put them in but still no luck. My primary concern is I cannot get the spa to even acknowledge that the pstn line is ringing on an incoming call. I here the call about wiring, but its very simple: I plug an analogue phone into my wall socket, i ring the number from my mobile, the phone rings, i pick it up get speech path etc. I plug the SPA line port into my wall socket, i ring the number from my mobile, nothing happens, no lights flash, the debug log prints zilch, nada, nothing. If i refresh the system page while im getting ringback on the mobile, i see the voltage fluctuate a bit, typically, betwen -7 to +7 volts and anything in between, now that seems a lot on the low side to me ( i would expect 40V or so), but this is plugged into the master socket, with nothing else plugged in. short of sticking a voltmeter on the line what can i do? I believe i have added the correct UK settings to the FXO/international stuff, but this seems to be so basic its even beyond config. So I think that my SPA is broken basically, and I need a new one? PS: I now have line 1 to and from asterisk working ok so some of my configuration is working, but the PSTN line is dead in all directions. |
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| I'm pretty sure the *default* config on a 3102 in the US will let the PSTN ring through the Phone port. If it isn't even doing that much, your unit is dead. Based on what you've said about what you see on the info tab when it's ringing and not ringing, it still sounds like its dead. I'd try another unit. --Donnie |
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| OK Thanks, there is one more thing to try before i return it. Since i am in the UK, we have TIP/RING on wires 1/4 of the RJ11 connector, and a seperate "ringing" wire on 2. I read somewhere that the SPA is expecting TIP/RING on 2 and 3 on the RJ11 - american style. And therefore this could be the cause of the issue. It might be there is supposed to be some "converter" for use in the UK? Can anyone confirm/deny this? I did recieve a converter for the phone port -> analogue phone but this cant be used the other way. I am not been able to splice together two ends of a cable yet (none of my rj11 plugs have wire 3 crimped |
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| Thread | Thread Starter | Forum | Replies | Last Post |
| How do I configure My SPA3102 to transfer a VOIP call to the PSTN line | techtek | Linksys (Sipura) VoIP Support Forum | 0 | March 31st, 2007 12:20 AM |
| spa3102 with local pstn line and pbxes with multiple extensions | misterm | Linksys (Sipura) VoIP Support Forum | 1 | January 31st, 2007 11:23 PM |
| How to make PSTN busy while line 1 on VoIP call in SPA3102 | Tadeusz | Linksys (Sipura) VoIP Support Forum | 0 | November 3rd, 2006 03:58 PM |
| Can't pick up forwarded incoming PSTN call to PAP2 | hunok99 | Linksys (Sipura) VoIP Support Forum | 0 | November 1st, 2006 07:08 PM |
| VOIP>PSTN stopped to pick up the line | majofa | Linksys (Sipura) VoIP Support Forum | 3 | March 24th, 2005 02:52 AM |