similar to: Act-Tel G11112DS Telephony Gateway

Hello everyone. I had heard about this open-source PBX once a while back. I wasn't too interested in it at the time but I kept the info filed away for possible future use. A couple of days ago, I was walking around Barnes and Nobles and I found this book, called Asterisk: The Future of Telephony. I paged through it a little and I was really excited by what I read. Then I remembered the

I have asterisk running more or less ok but I would like to turn off call waiting and be selective about the incoming sip connections. This is running asterisk 1.2.8 with a fxs and fxo card and a configured voip (sip) line. Currently I'm using freePBX 2.1.1 to configure asterisk. Problem 1) if someone is on the phone already and another call comes in for an already engaged extension I

Has anyone on here used kiax before? I am asking because I have it installed on several computers and have been able to get it to connect and register to my Asterisk box. I can even call between them and my SIP softphones. The problem I am having is this: when I use kiax to call someone else, they get some kind of background music playing while I am talking to them. I have called from kiax to

Basically, I am looking to set up an extension which will be used as a "help-line". I want it to function kind of like the bat phone from the old Batman series, where Commissioner Gordon would pick up the extension in his office and it would ring the phone back at Wayne's mansion. Is there a way to duplicate this functionality with Asterisk? I just need asterisk to auto-dial an

I want to set up a test system at my house using a FXO card (one of the X100P cards) and a FXS device. I am going to have the one line running into the machine and the house phones running off of the FXS port. I will be installing a softphone on my laptop and will be using Hamachi to tunnel back to my Asterisk machine while I am out and about. This way, I can leave my firewall intact and still

I have gone through some archive about Nortel MICS (Meridian ?)+ Asterisk Integration but I'm not sure whether same as my case . 70 telephone sets | | Nortel MICS 4.1 --------- Asterisk | PSTN I have read the David Gomillion's Guide and got the idea . However, my plan is slightly different from what he did , I need to use Nortel MICS to connect to PSTN (I

I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten => s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the

Thanks Rich, I have an SPA-3000 laying around, so I will attempt to set it up in a little more conventional manner (although your method looks like a winner for a home test PBX). Would you mind posting or PM your current config to me, maybe screenshots if you PM. If I start with that it will take less time to get to the point where the SPA-3000 is a true FXO-FXS gateway for *. I will be happy to

Voxilla.com has a great config wizard for the SPA-3000 and * http://voxilla.com/spa3kasterisk.php I took the output from this wizard and dumped it on my test box with an SPA 3000 (with some mods to match my * contexts) and everything worked great. Calls from the PSTN to the spa3000 are routed to dialplan #8 on the spa3000, which dials * Both the FXO and FXS port register with * The SPA3000 is

Does anyone know a good (and stable) voip gateway product with 4 ports (2 fxo and 2 fxs), with the following requirements: * being able to connect analog phones to the FXS ports, and communicate over SIP with an REGISTRAR/PROXY server (SER in our case). * being able to connect the FXO port to local office PSTN network, and dial to that office pstn number and getting an internal dialtone, or

Asterisk handling My Skype Calls This is for me, once more, Asterisk as the Future of Telephony. Today I've integrated my Skype Account as SIP extension in my * Box. This has been possible using "Uplink Skype to SIP Adapter", available for free at http://www.nch.com.au/skypetosip/index.html . Main features that any one can easily integrate into Asterisk: - Route skype incoming

I have been looking around some and I can't seem to find anything which will answer my question. If I have two Asterisk boxes in different locations which are linked to each other over the internet, can I configure the boxes to use each other's lines as local? In other words, let's say Site A has Phone1 for a 1FB line going into it on an FXO port. Site B has Phone2 for a 1FB line

Many thanks for that info! > Peter > > One thing to consider if you only have 3 PSTN lines is the Sipura > SPA-3000 (you would need 3 of them, one for each line) > > We have 2 PSTN lines at our scout campsite, and they work very well, as > well as providing a simple power outage solution. > > They retail about ?80 + the VAT > > I can supply more information

Hi, every one I'd like to introduce some new feature of our products. mg3000-r fxo gateway provides more feature to work with asterisk. 1.play asterisk ivr with no interuption. when the mg3000-r received call from co line, it wouldn't conect instantly.instead, it start call to asterisk ivr first,when the ivr ready, it connect the co line. this feature make user feel friendly.

We are maxed out on our legacy PBX, and the question is the process of migrating to a new * system from the legacy. We current have 36 FXO lines coming into our site, and the usage on these lines indicates we can spare a few of them to launch the * server, and then move additional lines over as the * server/network/cable plant is built out and traffic moves to *. Eventually, we will want to

Hi all, I'm trying to install a TDM400P card, and I need some help. Please, see below... after dmesg command: [root@voip root]# dmesg via82cxxx: board #1 at 0xD800, IRQ 5 Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 3 for device 00:09.0 PCI: Sharing IRQ 3 with 00:10.1 Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1:

Hi all, I'm trying to install a TDM400P card, and I need some help. Please, see below... after dmesg command: [root@voip root]# dmesg via82cxxx: board #1 at 0xD800, IRQ 5 Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 3 for device 00:09.0 PCI: Sharing IRQ 3 with 00:10.1 Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1:

Hello! We are located in the UK, and we are planning to replace our old pbx with an asterisk based pbx. For outgoing calls our present pbx is connected to three PSTN lines which all have the same number. Internally, the pbx caters for quite a few extensions, and each extension can make outbound phone calls. Only very rarely does our call volume exceed three simultaneous connections (inside to

Can someone please help me with my config settings for my TDM12B. I have got the FXS port to work fine, but I cannot get asterisk to startup with any settings for the fxo ports. It appears to be something related to the message "Signalling requested is FXS Kewlstart but line is in FXO Kewlstart signalling". However, I have tried what appears to be every possible configuration option. I

I am interested. I'm using Callvantage and Centillium TA. I'm also using Asterisk with a X100P Card. Eventually I want to replace this with my Linksys NSLU2. I'm having mystery power issues with my linux server. Currently I have calls from soft and ip phones going to asterisk then out to the ta fxo to fxs. Regular pots phones go to a telephone distribution panel with the