similar to: Sipura SPA-2000 and Asterisk

I figured out what was going on with the lack of/stuck on stuttered dial tone. Apparently, there are two voicemail directories being referenced: /var/spool/asterisk/voicemail/default, and /var/spool/asterisk/voicemail/local. The sip phones were using /var/spool/asterisk/voicemail/local to dump VM messages into, yet the MWI looks at /var/spool/asterisk/voicemail/default. Does anyone know why

I have a Sipura SPA2000 2 line SIP FXS box with line 1 on port 5060 and line 2 on 5061. The SPA2000 is on IP address 192.168.17.6, and the asterisk box is on 102.168.17.2. Both SPA2000 ports(5060 and 5061) share the same IP address. Every minute I repeatedly get the following output: SIP Debugging Enabled 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.17.6 SIP/2.0 Via:

You can even set it to zero. Mine works well when in zero. The line pick up immediately :> -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Stenton Sent: Thursday, February 09, 2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN

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

Greetings, We are currently testing a Sipura SPA-3000 as a gateway from our Asterisk system to a PSTN line for 911 access. We have a number of locations and want to place an SPA-3000 in each, connected to a PSTN line that will provide the correct ANI/ALI information to 911 for each location. It all works great, except for a reasonably significant (4 seconds) delay between when the SPA-3000

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

<< 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

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

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

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+

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, I am very new to asterisk and voip in general and so far have managed to get the FXO card and a few sip phones working fine. My question is where does the Sipura SPA 2000 come in the picture? Can it be used as an extension (i.e FXS) ? Or is it to be used as a line (i.e FXO)? Or it can be used as both? My understanding is that its just like another ATA186. Is that true? I guess what I

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

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

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]

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'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

I just got a refurb Sipura SPA-2000 and was able to assign it an IP address with DHCP and ping the device, but then I ran the firmware upgrade utility to bring it up to spa2k-2.0.13g which seemed to work just fine, but after it rebooted I cannot connect to its webserver for configuration. I can still ping the unit. When I use the built in voice menu it reads back the right IP address, webserver

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,

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....