similar to: Sipura SPA-1001: Bad Outgoing Call Quality

Hi everyone, I just got a Sipura SPA-1001 and have connected my Tivo Series 1 (yes its old). When I do a test call with Tivo, the call always fails (it seems to dial the number but never connects). I can pick up the phone line and hear the Tivo "talking". I've tried looking around for anything special I need to do but its still not working. I can connect a phone to the SPA-1001

I hooked up a SPA-1001 with asterisk yesterday and all works well except the phone doesn't ring. The phone I'm using has a LCD display so I can see the call come in. (with caller id info) I can answer and complete the call but it's just not ringing. The phone rings if pluged into a POTS line so it's not the phone that's the problem. I've used the SPA-1001's web

Hello, I've been experimenting with Asterisk 1.0.6 and a Sipura SPA-3000, and I've run into a couple of questions I haven't yet found clear answers to: It appears that the SPA-3000 has no call progress on it's FXO interface? Asterisk considers a dial() as answered when the SPA-3000 has dialed the number on the PSTN line, not when someone has answered a phone on the

Hi All, I'm on of those brave souls who bought into the preproduction beta of the Sipura SPA-3000 FXS/FXO adapter. I've had the unit a few days and am exploring it's workings. I really want it mostly as a straightforward FXO adapter, to replace an X101p. Let me be clear, I'd love to support Digium in every way possibe, and will likely buy a TDM40 card shortly. But, the X101p has

hello my friend is trying to get his spa-1001 (sipura) 1001 connected to my asterisk box. he reset his spa-1001 to factory defaults I emailed him the voip-info page I found on google and yes I did look on google anyways he isn't able to get the thing to connect to it eg getting a dial tone, he did install x-lite and it worked fine with that am running asterisk@home 1.0 can some one please tell

Hi, I'm looking for a full list of xml provisioning variables of the SPA-2100/3000. Currently the Sipura website has example XMLs only for the SPA-841 [1] and SPA-941 [2]. I'm mostly interested in the CallerID type selector variables and whatever variables control the PSTN<->VoIP settings. Sipura Configuration website form field names are numeral only. :( [1]

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

Hi all, Anyone able to remotely reboot their password protected Sipura SPA-3000 from command line. I am trying: Sipura SPA-3000 from command line: # wget http://admin:mypassword@192.168.1.55/admin/reboot The strange thing is it works fine when I go to http://admin:mypassword@192.168.1.55/admin/reboot with my web browser... Thanks....

Sorry to bug all of you with this, but I have to assume there are a number of Sipura experts here... I have a Sipura SPA 2000 that I've been using for nearly 2 years now. It's flashed up to firmware 3.1.5. On line 1, I no longer get Caller ID (it used to work, and I can't remember when it stopped). On line 2, I always get Caller ID. To my old eyes, _every_ switch on both lines

Hi, I'm using the SPA-2000 with firmware 1.06 on the Asterisk PBX, which works great for taking and placing calls, but for for some reason I can't seem to clear the stutter dialtone by either calling the extension I'm on, or the voicemail system on the Asterisk PBX. If I call my voicemail access extension directly, It tells me I have no messages waiting, yet when I hang up, then

I'm experiencing a problem with a Sipura SPA-941 not available for incoming calls after Asterisk & Freepbx upgrade. I can dial out with the phone gto any other internal or external ext. It is registered with the Asterisk server. When I dial the Sipura directly from any other extension, it goes directly to vm. I have other Sip softphones that are working fine. A sip debug when calling the

<< Has anyone seen something like that and is there a fix? A google search didn't turn up any apparent hits.>> I have seen exactly this problem. Even IVR failed to work. Got an RMA from the supplier and they exchanged with no questions. Bill Seddon -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On

Hi, Has anyone managed to create a setup with a Sipura SPA-1001 as a client? Right now I can connect to the device by dialing the extension number but when I try to connect from the phone handset to make an outbound call it gives an unavailable tone. I'm using Line 2 on the SPA-1001 to connect to the local asterisk server, line 1 is used to connect to my VOIP provider until I can get the

I want to implement a one way announcement and paging facility using Asterisk and Sipura phones. The wiki says Sipura phones only support Auto Answer using the Call-Info header which is no lone shipped with asterisk stable since 1.0.4. I would like to ask if anyone has implemented a similiar facility using Sipura SPA-841 or any other SIP phones. If I could take a look at how

Hi All, I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and out of Asterisk. I am using "PSTN Ring Thru Line 1" (on the "PSTN Line" tab) so Asterisk answers the call rather than the SPA-3000. It is all working perfectly except I can't get the SPA-3000 to pass caller ID to Asterisk. It passes "Display Name", "User ID" and any "PSTN

I'm still unable to get my SPA-1001 to work behind NAT with an Asterisk box out on the Internet. It works fine to my local A@H box. I've tried... many things. I'm willing to pay a $250 bounty (PayPal preferred) to anyone who can help me get it working. Any Sipura experts out there? Eric.

Hello all, So i am new to asterisk and very green when it comes to Linux, so don't beat on me too bad :) I just set up * on Red Hat 9.0 last night... everything "seems" to be configured coffectly, I can start * no problem and get the CLI prompt... now here is my question... I have an account set up with VoicePulse Connect! and also have a Sipura Spa-2000... i am trying to get make

Hello. Has anyone run into problems accessing voicemail with the Sipura SPA-2002's? Our SPA-2000's work fine (registers fine, able to make and receive calls properly & also able to access voicemail). We've configured the 2002's exactly the same way. However, with the SPA-2002 we're unable to access the voicemail system (though it does register fine and is able to

I had a Grandstream 286 at my home hitting my Asterisk box at the office, all worked well and I received phone calls fine until the device just up and died. I replaced this unit with an SPA-2000 because I have been impressed with the Sipura devices and decided to use them for most of my needs in the future. Problem is that my phone attached to the device rings shortly after power up of the

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,