similar to: SPA-2000 and time of day

I'm looking to give the SPA-3000 a whirl as I'm having too much difficulty with the irq sharing thing inside the box. I'm reading the book but without having one in-hand to play with it appears a little obtuse at this time. Before I drop down my money can someone with some hands-on with one of these confirm if the SPA-3000 can: a) detect inbound distinctive ring (this looks to me like

I upgraded my 7960 to sip v 6.3 and my display time has now disappeared from the top left corner. Loadid: SW: P0S3-06-3-00 ARM: PAS3ARM1 Boot: PC03M030 DSP: PS03AT38 Here is the section dealing with time in my SIPDefault.cnf file. Does anybody see anything wrong with it or have any other ideas? # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)

I am still haveing problems (echo) with my X100P but I'm thinking it has more to do with the server it is in which is not a negotiable item at this time. My question then is to the use of SPA3000's as a replacement from the FXO standpoint. 1. Can you setup the FXO port to recognize distinctinve ring and call a different context like you can do with Zap channels? Being able to call a

My cell vm goes to asterisk, not the carrier. Apparently MWI is turned on/off with specially formatted SMS messages. Anyone know how to do this on a Treo 600? Having the phone light from Asterisk would be HUGE ... not to mention extremely cool. dbc.

Asterisk as a phone system makes perfect sense in a condo. You can get all the DID's you want and eliminate costs for the owners. You can offer standard FXO for people who don't care and IP sets for people who want to "upgrade" to feature sets. Your door openner is a piece of cake. 1. Create an option in your dialplan only in the "from-access-door" context that reads

The machine in question is a DL360 G3 R03 Xeon DP 3060-512/533 (part number: 322471-421). It's the first HP-branded model we've had so far (previously all Compaq branded). It has 1Gb of memory and a couple of drives on the on-board SCSI controller (setup for mirroring in the SCSI bios). No SMP is setup right now (doesn't get far enough to do anything with it!). In any case, this

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

Did the console "Dial" command go away in recent releases? I just went to use is on my server (now at 1.0) and its gone. Help from the CLI doen't show it. -- David Cook

It looks like this meetup group is becoming the venue for contact as most "Toronto" respondents have signed up here. Can I request that the remainder who responded to the "Toronto" call signup? Shidan is the organizer and has proposed a date. I'd hate to confirm it without the remainder having input - especially those with travel times like Andrew K. out in Listowel.

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

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

> Lito Lampitoc wrote: > > thanks for enlightening. So you mean, if I have 3 lines when the > caller > > dialled the first line and it was busy, the call will be diverted > to the > > next two available lines in random? > > > > I don't think it's random. I think its just sequential. If main > line > is busy, try second. If that is

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 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,

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

Hello: Does anyone know how to configure any of the Cisco 79XX phones to get custom ringers when using chan_sccp with Asterisk? I've currently got Asterisk's 05-24-04 CVS-HEAD and Zozo's 0.2 release of chan_sccp. I've tried using ringlist.dat, but that appears to only be for the SIP phones... Thanks for any input, Andrew

Hello all, Downloaded, compiled and installed Asterisk CVS-04/15/04-17:54:5. Everything looks fine except I see a pseudo channel in the 'zap show channels'. xxxx*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo default 1 default The result is, I cant use Zap/g1 in extensions, eg this doesnt work anymore: exten =>

Hi I have found in http://lists.digium.com/pipermail/asterisk-users/2004-July/056111.html (Hack to make * -> (H323) -> CCM -> IOS GW work) that i need a special version of chan_h323, because of the External RTP problem. Do you know exactly which version is it? Or do i need an unofficial patch? Thanx Andr?s

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+

When I use an analog phone connected to a Sipura SPA-2000 it takes about 3-4 seconds before the number is actually dialled. Very annoying especially if you are connecting an intercom to it. Can I change this behaviour and do I need to look at * config or the config of the SPA-2000? Thanks!