similar to: Spa3000 doesn't hangup after a conversation

Hi, I have a little situation here :( Perhaps somebody can give me a hand with it. I have an Asterisk working, and in another office, a Sipura SPA-3000. I configured the SPA and I have the extension working, the incomming trunk working, but the outgoing trunk (peer) does not work. The issue is that I have a dynamic IP where the SPA is, and neither the SPA nor

Hi, I'm using a Sipura SPA 3000 to receive calls from a PSTN line and to send them to Asterisk. But on Asterisk, I don't see the PSTN caller ID! I only have User ID of the SPA 3000 as caller number. The caller number is present on the PSTN line. I'm in france, maybe the SPA 3000 is not compatible? If you have an idea, please tell me! Thanks, Micka?l

Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank

Hi, I have a little situation here :( I have an Asterisk working, and in another office, a Sipura SPA-3000. I configured the SPA and I have the extension working, the incomming trunk working, but the outgoing trunk (peer) does not work. The issue is that I have a dynamic IP where the SPA is, and neither the SPA nor my router have DynamicDNS. So, if I

Hey, all... If this is too off-topic, I'd be grateful for directions to a more appropriate mailing list. I'm trying to set up Asterisk and some Sipura boxes. I've got an SPA-3000 which is registering twice with Asterisk - once for its FXS/Line1/VoIP1 and once for its FXO/PSTN/VoIP2. My eventual goal is to have inbound calls on its FXO ring four times on its FXS and then fail over to

My porblem is incoing PSTN calls are being forwarde to the * box, the phone rings, but when the phone is picked up, the call is not taken - it continues to ring. I am forwarding the call to (<S0:6666>) in my dial plan Can anyone assist? This is driving my crazy! Extract from the * console Executing Dial("SIP/3001-047c", "SIP/2004") in new stack -- Called 2004 --

Hi, These days I run into something quite odd. I have an A@H that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a

Hi List, I hope this setup must be done by our astersik users.. I am using Sipura 3000 to receive PSTN calls and forward those calls to asterisk for voice processing and after that, I am transferring call to extension through FXS port on SPA 3000. Currently, media of call is trombone through asterisk. i.e achieving blind transfers on asterisk with SPA 3000. Is it possible to stop trombone

Hi all, I was wondering whether anybody here would help me clarify this minor issue please. If I have the following setup; Asterisk ------ Sipura SPA 3000 (fxo) --------- Pstn Line Would a call coming in on the pstn line be answered by the ATA or just get passed through to the * server (depending on dialplan) to handle? So basically, the caller does

Hello, Before throwing in the towel with my Sipura 3000 has anyone had much success with that adapter connected to a door phone? In our setup a doorphone is connected to the SPA's fxs port. When a visitor rings, asterisk calls a group of Polycoms and the person who answers has to enter *1 to trigger the door opening. However it seems the SPA doesn't relay the DTMF's to the

I've been running with a generic X100P for 5 or so months and every once in a while I have problem receiving faxes. I see that others have the same problems and some worse than I have with these boards so I was wondering if using a Sipura SPA-3000 would be any more reliable. Has anyone had enough experience to tell me if that would definitely fix the random fax error. PS. I have * at

Hi, all I am trying to figure out how to forward incoming PSTN call on SPA3000 to VoIP extension(s). Basically, I have converted my home to VoIP. I have normal phone (connected to SPA3000) and couple of IP phones. All call coming from VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I need to do same thing for incoming PSTN calls. I have enabled gateway function in SPA3000 and

The party's over folks, the new official cisco/linksys/sipura policy is to no longer sell SPA-3000's to end users. Buy them while you still can :-( -Dan

My apologies for the off-topic post ... No matter what settings I try, when I dial in to the SPA-3000 on the PSTN line, it picks up the call and immediately gives me a fast busy tone then hangs up. The info tab says under PSTN Line status: Last PSTN Disconnect Reason: PSTN Disconnect Tone which seems to indicate that the SPA thinks the caller has hung up. Since I am in Japan, it is possible

I am located in the UK, and I am using Sipura spa-2000 adapters to connect analog phones to a voip network. The network connects to the PSTN as well via the Sipura spa-3000 adapter. I would like to provide surge protection for the spa-2000 and the spa-3000 adapters. 1. For spa-2000, fxs port: What is the maximum tip-to-ring voltage before damage to the the adapter occurs? 2. For spa-2000,

I'm about to purchase an adaptor for a POTS data modem and was looking at the Sipura line of adaptors (SPA-1000, SPA-1001, SPA-2000, SPA-3000). Do these work well? Anyone have a suggestion on which model of the Sipura I should get? Does one work better with * than the others? Are there other adaptors that work better that I should get? Thanks, -Dave -------------- next part

Dear list, I am playing with Sipura 3000 since last week. Through the wiki pages I could get running it reasonably well. My setup is that of a Sipura, linked with a local analog cordless phone, a local PSTN line and the setup to link to an asterisk server located at a remote static ip address. I can dial the cordless phone from other extensions located at the asterisk server; I can dial out

Hi there everyone! I've bought a Sipura SPA 3000, and succesfully connected it to my Mac, where I installed Asterisk 1.4.0. Both ports (FXO and FXS) are well configured). However, living in Brazil, I'd like to know if there are optimal settings to my PSTN that I should enter into the config of the device. I experience a little bit of echo on the FXO probably because I raised the gain of

Hello all ! Can anyone recommend me ATA device that REALLY has T.38 built in. So far I have heard of Telco Systems Access201, which seems to be impossible to bye in Europe (all resselers are droped Telco systems ATAs for some reason (tried in Germany and in UK so far)), and I have heard that SIPURA SPA-2100 should have T.38 built in into newer firmware, but I wasn't able to confirm that

Some people on this group may have understood from messages posted here that the Sipura SPA-2000 is not currently available for shipping. That is not the case. Voxilla.com has the Sipura SPA-2000 available for immediate shipping, and has had them since late November. The price is $109.95, and it comes with a month of free VoicePulse service with activation fees waived (a $65 value). In return