similar to: Asterisk and Sipura SPA-841 SIP phones

I am looking at some lower cost phone to use with Asterisk. What is the ArtDio IPF-2000 or the Sipura SPA-841 like? Also, I see voipsupply.com has an ArtDio IPF-1000 listed, is this a new or an old model? I cannot find any information on it. Adi

Hei! Does anyone know how to configure this phone to autodial the number after interdigit timeout has passed? Rennes

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

Froogle found me one supplier for the SPA-841, not sure I trust them though. Does this phone even exist yet? Does anyone have any experience with it? Does anyone know a vendor other than Atacomm/voipsupply?

I have just bought several Sipura SPA-841 SIP phones, and after some testing I have found out that the volume received by other parties when calling using the handset is very low. I've been able to reproduce this problem in the 3 phones I've tested so far. I've tried tweaking several configuration options but nothing I has helped so far. Has anybody else experienced this problem?

Anyone having problems with registration to * from a SPA-841? I got a SPA-841 a week ago. I noticed that sometime it could not be reached (dialed to) and it can't dial. In this case the line LED is yellow. I enabled logging to syslog and there is a hint as to what happens. For some reason sometimes it gets "401 Unauthorized" Any ideas what is happening and how to fix it? Phone

Hello, How good is :SPA 841 form SIPURA. Thanks Varun

Hi all, The problem is on the volume of the voice sent by the SPA-841. I think the echo cancel algorithm sets a limit to the microphone when detects sounds or noise from the earphone. This problem generates an oscillation on the voice volume sent by the phone and even turns it off completely for very little lapses of time making the communication very uncomfortable. I manage three different

Sipura has implemented auto-answer in version 0.9.5 of the SPA-841 firmware. However, it is implemented via the Call-Info header, which Asterisk stable doesn't currently support. The attached patch implments a quick hack to support the Call-Info header from the Dial() application by way of setting the CALL_INFO variable. For example, the following macro can be used to dial up a single

I heard Sipura had really awesome documentation on the SPA-2000 and SPA-3000, but you have to email them for it. When I did, they said I had to get it from a reseller. It's been a while since I bought my units, I don't even remember where or who they were bought from. Can somebody email me the documentation for these devices? I'm quite interested in knowing what every one of their 200+

If I understand correctly the Sipura people are the same guys that made the Cisco ATA (Komodo phone) or what ever. I'm going to get one of the Sipura SPA-2000 to use and abuse with *.... I have seen the web interface.. John over at Chagres was nice enough to let me login to one and look around a few weeks back... I'm impressed .. if you guys care to buy one

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

Anyone have one of these yet? http://www.voxilla.com/shop/index.php?action=item&id=38 -Dan

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

Anyone had any experience here on how to config both ends, asterisk and the sipura SPA1000 TIA

I recently upgraded from a few month old CVS version of Asterisk to v1.0.1, and dialing out through my SPA-3000 stopped working. Notice right after INVITE, in the old CVS version, it includes the number I'm trying to dial (8019596) which works fine, however in v1.0.1, it doesn't include the number and of course the dial fails. Did a config option change out from underneath me or

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

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

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

Hi, the topic says it all really. Does the Sipura 3000 detect and report UK clid correctly? thanks Mike