similar to: Pricing for DS3000P

Hi All, We are trying to implement a DS3 capacity calls (672 concurrent calls) using asterisk server. I wanted to ask are there any compatible DS3 cards with asterisk? I tried searching a lot but could find DS3000P from digium but unable to get this product. Does anybody have any idea of having any DS3 card in asterisk box so as to handle around 600 calls? Thanks Sandesh -------------- next part

http://www.imagestream.com/PCI_720.html Regards, Andrew

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

Do you remember when Digium annouced channelized DS3???...vapour. You watch, this CODEC board will go the way of the DS3 boards...vapour. Mark my words. --------------------------------- Yahoo! Messenger with Voice. PC-to-Phone calls for ridiculously low rates. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Everyone, I must deploy an asterisk system that can support at least 500 pstn outbound calls. It's a challenge as it's the first time i'm gonna build such a large system. I want to have your advice on hardware, software and so on . What i have in my plan is a cluster of servers with quad PRI cards. I will appreciate your advice. Thank you all . --

I'm trying to find the right Digium card for the Dell 2950 Dell 2850 HP DL380 G3 HP DL360 G3 Are these 3.3v or 5.0v machines ? I am out of the office, and need to buy a card today. I am looking at either the TE407 or TE412, and would appreciate any help. :) Julian

Obviously, there are no DS3 TDM cards that are currently compatible with Zap channels. (or are there?) Does anyone know of an inexpensive DS3 card that could perhaps be used with Asterisk if one were to try to port the Zap drivers to such a card? PCI, of course, would be the bus of choice. I think there are quite a few discouraging comments to be made on that question. Firstly, most

Hi, I'm planning to get my Asterisk box out of the LAN, get rid of my router and make the box acts as a Router, Firewall, DHCP Server (with Shorewall). I'll do that to be able to use some SIP clients remotely. Does anyone doing the same with the Asterisk box, is it a good idea, is there any other solution for the SIP emote Clients. Regards. __________________________________

Hi, Does anybody has experience with high capacity PSTN voicemail and asterisk, running more then 5k mailboxes for PSTN users ? How many mailboxes can I serve with 4xE1 card if we assume that we have enough harddrive capacity. What would be server requirements. Would the CPU load be the same when storing voicemails in gsm format as compresing to gsm for ip calls ? Any hints would be greatly

I don't want to start the "How many calls can Asterisk handle?" discussion or "How many angels can stand on the point of a pin?" discussion either. But can anyone contribute some practical knowledge of systems that take in channel bank T1s or DS3s from "far away", and process the calls? I am looking for real world, been there, done that, or "check the

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

Hello Everyone, We are looking to interconnect with a local ILEC over an OC-n transport layer. They basically gave us two options in terms of mapping the SONET to the DS3: * VT1.5s mapping * DS1s mapping The second option is quite clear. We would MUX the connection, and plug the lines into qaud t1 cads etc... The tech mentioned that with the second option we would also need a DACS to convert

Hi, I find this diagram which details the kernel packet traveling : http://www.docum.org/docum.org/kptd/ Is it up to date ? I made some test and I put a DNAT rules in the PREROUTING table of an interface and I attach it a ingress policy, the dst IP wasn''t changed. the DNAT it isn''t yet make. I''ve another question (I''m not sure is it the good mailing list), for

Any know what Digium hasn't released the DS3 card? It was supposed to be out a while ago. -Matt

BJ, >BJ Weschke <bweschke@gmail.com> >Subject: Re: [Asterisk-Users] Do Both! :) Re: Telecom >SIP termination vs. DS3 >To: Asterisk Users Mailing List - Non-Commercial >Discussion <asterisk-users@lists.digium.com> >Message-ID: <79cf63305051908056c284cc9@mail.gmail.com> >Content-Type: text/plain; charset=ISO-8859-1 >Did I miss pricing/availability

http://rgd.legalaffairs.gov.tt Test User username: izo password: izo May be a bit heavy on the graphics by I am still try to teach the graphic artist about web 2.0. :-) regards, Leon Leslie -- Never be afraid to try something new. Remember, amateurs built the ark; professionals built the Titanic! --~--~---------~--~----~------------~-------~--~----~ You received this message because you

All This is a small issue that I stumbled onto that has to do with the channel numbering on an E1 connection into an Asterisk Zaptel/DAHDI system. As most of us already know an E1 has 32 channels of which 30(1-15 17-31) are B-channels and 1 (16) is a D-Channel. The 32nd channel is not presented in Asterisk Zaptel/DAHDI. There are other configurations but this is the most common. ***

Message: 16 Date: Thu, 19 May 2005 00:16:34 -0600 Michael, Do both! As for Sip Termination: ----------------------- Contact Kristi Eggers @ Txlink.net for month to month Originating/Termination VoIP Toll Free or Local USA DID #s. Yes they do both Sip and IAX. You must have seperate accounts for either Sip or IAX and fund your account with a minimum of $100. This is what I did. Once I get

Dear Rails List, I am new on the list so will assume straight away that this question has been answered before. As I can see no immediate way of searching the list, I will ask the question. I apologize in advance for any disruption this may cause. Question time: I am trying to write a web application for the staff at my office to play a form of virtual Super14. Its a bit like any of the

Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: