similar to: Get SPA-2000 to dial out on one * and get calls in from a different *?

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 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

In my hunt to track down my echo issues, I tried disabling all echo cancellation, suppression, adaption, on my SPA-2000 (Advanced section of the config, under Line 1/2). Then calling from one local extension to another. (SPA-2000 Line1, to Line2 on the same device) I was pretty shocked with the results, the echo was HORRIBLE! I even tried 3 different analog phones. Now, once I turned the echo

Hello All! I am in the process of assembling an asterisk-based phone system for my office - using SPA-3000s to connect the network to the PSTN. I am wondering if anybody else can get (or has already seen) the same behaviour out of their 3000. The short version: Send multiple Calls to the SPA's FXO port at the same time it is re-registering with Asterisk. SPA HTTP Configuration:

hello my friend has the proxy set up his extention set up his password set up but he isn't getting a dial tone is there a second setting we need to put the address in? he is going to advenced settings line1 and in the proxy address box he is putting the info in below is the way he has it set up Sipura SPA Configuration Sipura Technology Inc Info System SIP Provisioning Regional Line 1 User 1

I am having trouble getting transfers to work when a zap channel is part of the call. I have a couple SPA-2000's and some X100P cards as my setup. This is what I'm trying: Dial number from phone: -- Executing Dial("SIP/206-2c61", "Zap/1/#######") in new stack Currently on call: -- Called 1/####### Press flash to place call on hold with SPA-2000: --

I have one analog line hooked in my Asterisk box using an x100p (I think that's the model number). When I do this in my extensions.conf: exten => 1200,1,playback(pls-wait-connect-call) exten => 1200,2,Dial(Zap/1/5555551212,20,rTt) exten => 1200,3,VoiceMail(u100@lightwavetech.com) exten => 1200,4,Goto,t|1 The phone rings beyond the 20 second timeout and never really goes to the *

Hi list, After successfully configuring Linksys SPA3000 and SPA3102 devices as Asterisk PSTN gateways, the only thing I can't get working is the PSTN Caller ID. The analog and SIP phones I've used can both display CIDs for internal calls, while the analog model also displays CIDs correctly when attached directly to the PSTN line. However, when PSTN calls come in via the SPA

Hi All, I have a asterisk box that is now on its own static address on the net.it was originally behind a nat firewall. The problem I have is that the remote SPA-2000's that are behind nat firewalls now fail. here is relevent sip.con entry [2001] type=friend username=2001 host=dynamic defaultip=81.178.77.67 allow=ulaw dtmfmode=rfc2833 mailbox=2001@local context=sip

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

Hi, I have not implemented any of the spa-2000's yet. Do they work ok with asterisk? Is the 2000 capable of having 2 FXS extensions off each one or is it two fxs ports with the same extension?

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

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

Hello, I am trying to build up a pretty meaty Asterisk box after doing our initial testing and playing on a 1ghz system. Right now I have decided on a prebuilt system which I normally don't do but thought it seemed like a good deal. I have included the initial specs below, I will be adding another 1 GB of RAM for a total of 2 GB. My first question is regarding the serial ATA drives... I

Been having a problem with my two Sipura 2000's dropping calls from the SPA-2000 side. Seems the calls are dropped right before the "Next Registration" time. Calls drop about ever 60 minutes or so. I have dialed from one port to the other and let it sit. After about 60 minutes or so the calls get dropped. System details are below Asterisk ver. CVS-HEAD-11/27/04-23:42:45 RHEL 3

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've been trying to get fax reception to work using an SPA-2000 to ring the fax machine or modem that's taking fax calls. I was curious if anyone else had tried something similar, and if so, had any luck getting it to work reliably. I've been able to get it to work, but it isn't reliable. (Pages/lines of black dots result more frequently than not.) The incoming lines are FXO

Hello all, I have a new Sipura SPA-2000 that I am trying to configure beind a NAT. The SPA is able to register to the asterisk server without a problem and the SPA can make calls to other extension that are not behind a NAT. However, when I try to call the SPA from another extension, the extension connected to the SPA rings, the user at the SPA answers, and there is no audio in either

I started out a few years ago with some SPA-841 sets, because the Grandstream 2000 I thought I wanted was perpetually delayed. The GS had more call appearances, and I didn't want just the 4 max that the SPA offered. As it turns out, with the greater flexibility of VOIP, I don't need 'dedicated' CAs the way I needed them on ISDN previously, so 4 is actually adequate. Along the line,

We have issues with all of the SPA-2000 ATAs we have where incoming calls from only one of our Asterisk servers do not complete. Details: 1- On the CLI we see that when the call is pushed to the ATA it shows Busy/Congested 2- We can make calls to this same server just fine 3- We can receive calls from other Asterisk servers running older CVS versions of Asterisk with the same exact ATA