similar to: Can this box handle 8 T1s (PSTN) with Asterisk?

I have compiled zaptel on Mandrake following everything I have always done on Fedora. It is 2.6 udev so... I had to modify the 01-devfs.rules Make linux26 Make Make install... Everything appears to compile correctly but it says module not found when doing "modprobe zaptel" Is this a rights issue? Jordan Novak -------------- next part -------------- An HTML attachment was

Please help me test my setup by dialing 800.564.0215 and listen to the queue for a bit. I have a quad port T1 with NFAS setup. I can dial-out but I cannot dial any 800 numbers (Global Crossing says I need LDS service and that will be a couple weeks) so I cant test it myself. I need at least 24 callers to feel comfortable enough that it is working properly. Thanks, Steve Totaro

Hello all, We are looking for some hardware requirements/recommendations to be able to handle a full DS3's worth of TDM -> VoIP traffic. The DS3 would bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then need to convert those calls into G729 SIP VoIP calls to send to our asterisk box over ethernet. Since everything is going in/out of asterisk is 729, and no features

Hi, I am investigating getting a wifi VoIP phone because its may be a better option than an ATA and a cordless phone.. Does anyone have any experience with the whats out there?? Do they support things like WPA etc?? I have heard the battery life can be a problem.. Is this the case? Thanks..

Hi listers, I am looking for people who have used Power over Ethernet switches, primarily in conjunction with Polycom IP 501's. I've been looking at the Linksys SRW224P, since I've had good luck with the SRW224 in our office. However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary in price. I would appreciate any input people have to offer. Thanks, James

Has anyone got this annoying sidecar to illuminate when users are on the phone? In my function key settings I have: Context: Active Type: Extension Number: <sip:4000@serve.address.com;user=phone> (4000 is the extension I want to see/dial on the key). I can press the key and it will dial the extension, it just won't illuminate when the user is on the phone or on DND Since I have

Thanks everyone for the input. After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over. I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no home phone (just cell phones) so this'll be a great way of testing. All

I have a client that has 5 Snom 320s. 4 work great, one does not. I upgrade the firmware to the latest (6.5.2) and the problem goes away, but then comes back a couple days later. There is a slight packet loss on the phone (about 1%), though there is no packet loss on any of the other phones. I determine the packet loss by the Linux command "ping -f -c 10000 192.168.2.10". Outgoing

Hello Everyone, We are getting some rather poor results (relative) with our Asterisk setup. Not sure if we are using the sipp correctly etc.. but nevertheless, is there any documentation that describes how we can get the most our of our Asterisk box. For example when we hit the "too many file" error, and fixing it using ulimit..... Also, is there any way we can allocate sufficient

Does anyone have a guesstimate of how many active Asterisk installations there are? Sorry this is off topic, need it for a customer proposal and they need comfort on stability. A count of the downloads from Digium would be a good start but I couldn't find this anywhere with Google. Feedback from anyone who may know near actual data would be appreciated rather than simply guessing,

Hi, When making a call from an Asterisk box over a PRI connection, I am able to set the Caller ID phone number to what ever I want. This works find. How to I make the called party callerid display "Confidential" or "unknown" as we sometimes see ? Andre

Does anyone know of a really lightweight web interface that allows users to log in and modify attributes of their extension only? Thanks Curt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061030/892a67b2/attachment.htm

Hi, I've just installed asterisk 1.2.1 on my openwrt distro ( I own a WRT54GL by Linksys ) . No problem by now, but I can see that my 3 snom 320, once they started they send subscriptions to asterisk, and I can see that running: sip show subscriptions But, after one hour about, OR when I do asterisk reload , asterisk losts all th snom subscriptions. Someone can help me please? Thanks

I am tasked with linking an Avaya Definity switch to an asterisk box using it's IP card that handles H.323. All my googles turn up a lot of results but nothing recent. I am able to find instructions but they are dated from 2005, and often fail halfway through. What is the best way to achieve what I want, which is two way calling between the Avaya switch and Asterisk server using h.323, and

Hi All, Just a little over thought. Sorry if someone already asked about this before. Is it possible to put all 16 Ports of E1 in One Asterisk Server ? And if it's not possible is there any suggestion or alternative for me to use more than 320 lines of outgoing calls on One Asterisk Server ? Thanks ZH -------------- next part -------------- An HTML attachment was

Hi all, general query here --- I'm about to set up an asterisk box for use in Japan but can't figureout if it's all ISDN there or what? I have gathered so far that the two major providers, NTT and KVH both offer ISDN lines with .......INS1500 and maybe INS64 protocols? Not sure... But I'm seeing stuff about J1 vs. T1/E1 .... so does that mean I can't use a Digium card it

Without using Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ / \ User2 ------ Go to register ------- SIP2 ----- Whereis? --> DB / \ / User3 --

We have a number of clients who will be needing a server to host Asterisk on. Many of these clients use analog (FXO) lines that will need to be connected to Asterisk via Sangoma cards. Can anyone recommend an industry-standard server (like IBM, Dell, HP, etc.) that has enough open PCI slots to handle up to six of the Sangoma cards? We would like to be able to tell the customer to just go

Good Afternoon, I have an Asterisk box that is acting like it is passing "core show" before every command I type. For example, if I type sip, I will get "No such command 'sip' (type 'core show help sip' for other possible commands). Any ideas? -- -jayson

Hello, I run a small network consulting company in the Chicago area and I have a client who is interested in doing an asterisk based VOIP installation. My company does not have the necessary experience to carry out the project alone so I am looking for an asterisk guru to lead the project. I'm interested in someone from the Chicago or northwest Indiana area who is very experienced with