similar to: SPA-962 Time on Asterisk

Good Day, I have been tasked with fixing the time on our asterisk server. I am having a hard time finding documentation to tell my what asterisk uses to get its time information to push to phones (or a better question, where does the SPA-962 get its time information)? Basically, I can go under the settings of the phone and change the offset to set the correct hour, but it is still about 4

The receptionist at one of our offices called today to stay that there is an extension on the SPA-932 console that has been solid red for a few days now. He has restarted the console and we restarted the phone in question. Still the light stays solid red despite the fact that there are no calls in progress. Since this isn't happening on any other extension I am not sure what to look at.

Good day to all-- First off let me say that I have been very pleased with the mailing list. I have learned a ton of stuff just reading other peoples questions and comments. I really enjoyed the VOIP Conference call on Friday morning. Still working on figuring out the best approach to custom voicemail emails (the reason I joined this group); however, we have more pressing issues. I

Kevin Walsh noted that his SPA-2000 takes time from his local NTP server in a post back on Fri June 25. Q: Where do you tell it to use NTP? I'm a bit confused as to where my SPA-2000 is currently getting its time. I told it GMT-5 in the misc section but it doesn't really tell me where its going for this. Is it just broadcasting looking for ntp? The net of my problem is that it is 1 hour

Has anyone gotten call-waiting to function on the 4 line SPA-841? I've seen some documents that say it can do it, some say no way. If yes, can you share configs / SPA-841 settings? If no, did you work around it? I can call out just fine on all 4 lines, however, if I am on the line, another call coming in does not ring the 2nd line...it just goes to busy / VM. -Darren

I just bough a Sipura SPA-2100 to use with Asterisk. When I use the analog handset plugged into the SPA-2100, the person on the other end can hardly hear me. I check the SPA-2100 setup and their is no mic/spk gain control. Is this a problem with the SPA-2100 or with Asterisk? Any way for asterisk to compensate for the poor audio level (if the problem is the SPA-2100)? Thanks, Mike

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 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

My SPA-3000 finally arrived and I'm trying to get the FXO port on it to work as if it was a X100P card as far as Asterisk is concerned. I have Asterisk dialing out over the SPA-3000 FXO port no problem. The issue I'm having problems with is having the SPA-3000 automatically forward all incoming PSTN calls to the Asterisk "mainmenu" context (or ext I guess). Currently the

Greetings, What is the recommended settings for using SPA-3000's FXO port for dialing out to PSTN in regard of the DTMF? The voip lan contains SPA-2100 and SPA-3000, with all fxs/fxo ports registered to the Asterisk box with unique username/passwords. The inbound PSTN DTMF works excellently, e.g. people calling from PSTN into the * box are able to pick IVR items with DTMF reliably. The

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

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, I have a Linksys SPA 3102 using as ATA. The call routing is : Phone -> PSTN -> SPA 3102 -> SIP Proxy -> Asterisk The problem I am having is that when the phone hangs up, SPA 3102 can't detect it and relay the CANCEL message. Is this problem with my SPA 3102 config or it just works like that by default? Thanks in advance for your help. Regards, Mark -------------- next

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 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 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

This post is simply documenting a spouse-friendly way of using the spa-3000 as both a fxs and fxo port for basic soho environments in the US, allowing asterisk to participate as needed/wanted. All home phones are connected _only_ to the spa-3000 fxs port. The incoming home pstn line is connected _only_ to the spa-3000 fxo port. Defined Line 1 (fxs) to register with asterisk via sip (extn

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

Hi. I am thinking about building an asterisk system for a small business and want to be able to page through the phones. It seems like to do this asterisk needs auto-answer support in the phone. I know the SPA-841's support this, as do Cisco phones, but I have been unable to determine if the new Cisco/Linksys SPA-941's do. Does anyone here have experience with trying to use auto-answer

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