similar to: Channelbanks for use in europe (Sweden)

Does anyone have any recommendations on how to connect 160 analogue phones to an asterisk PBX? Background information: A client wishes to replace their current PBX with a new VoIP system. Currently they have 2 PRIs. I intent to set up 2 asterisk PBXs with Debian GNU/Linux on raided drives. These drives will be mounted only read-only to recover gracefully from power-cycles. I am considering 2

Ok, the Zhone sucks and the Adtran 750/850 seems to be a little too expensive. Can anyone recommend a decent channelbank that won't break the bank? TIA, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year

Hi All. I'm working on an * configuration. We require 8 inbound POTS lines, and CT1 or PRI seems like it will be quite expensive at that level. I've read that a T1 Channelbank plus the T100P would be a (the?) way to go for this situation. What is the recommended channelbank for use in this scenario? From searching the archives I see a lot of suggestions to get "a

Trying to find my feet here. If I wanted to connect Asterisk to a PRI and throw in a T1 Adtran channel bank into the mix for fax machines would the following work? Connect PRI line from telco to Port 1 on the Digium Wildcard TE205P. Connect Adtran TA-624-T1 to Post 2 on the Digium Wildcard TE205P also, would I need a crossover to the channelbank or is it a patch lead like the connection to the

hello, I am planning a fairly large hotel VoIP system, using analog phones. It will consist of about 100 analog phones, that must have access to a VoIP server. I am considering an option to use a couple of asterisk boxes, bundled with a total of four TDM2460E cards, and one TDM2451E card. Has anyone on this list done something similar? It would be great to hear some comments regarding a smilar

Hi- I have posted a photo of the TE410P Digium card on my site, so that those wishing to purchase a compatible motherboard can see physically what the PCI slot requirement is: http://www.evtmedia.com/TE410P.htm I believe the required slot is a 64-bit, 3.3 Volt PCI, most commonly found on Xeon-based motherboards. A 64-bit slot is longer than the TE410P board requires, but the PCI connector

Using a TE410P with Zhone 24FXS channel banks to power standard analog phones I can't seem to find out if it's possible to support FSK or voltage type message waiting lamps. I don't want to use stutter dial tone because of the dramatic difference in per phone cost. TIA

Hello, Avaya has a nice feature that allows you to a) ring both a cellphone and a desktop phone at the same time b) transfer calls (and access other PBX features) from the cellphone that recieved the call, as long as the call is bridged through the PBX c) while talking on the cellphone, pick up the handset on your desktop phone and the call is automatically moved there, hanging up the cellphone

We are a voip terminating company, we're using Channelbank with FXS modules, Rhino, CAC, etc.. What we're wondering is, is how to would you echo cancel a channelbank. Of course we're realizing that cancel'ing on the T1 (on Ast) does no good (we think?) because the analog conversion is at the channelbank. Suggestions? Lowering the gain helps but we're looking for a real solution

Just a couple of questions for Aussie users/resellers! I have only just started to look at asterisk a couple of weeks ago and have found some very intresting discussions and some useful info on what can and can?t be done with it and technology. The questions i have is, Are there people using it is Aussie?( I would say yes so prob answered my own question ) :) Hardware, is the digium

hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. it will be a slow migration, the asterisk server will be inserted between the telco E1 and the hicom. new phones will be sip ones. the customer has several fax machines and analog phones (some of them have to be explosion-proof). around 50 analog ports in total are needed. as we are in

Hi all, We have been fighting with telco for a entire week. Today they came here with a LITE3000 to analyze what is going on. When I configure zaptel with no external clock, E1 gets aligned/synchronized with bit rate in 2048000 bps, both me and telco. span=4,0,0,ccs,hdb3,crc4 But when I configure span4 to get clock source from telco they become unsynchronized. TElco bit rate stays in

Hi List, After reading a bunch of the docs, list post archives, it still seems that a clear definition of how to clock the T100P card is muddy. zttool says that the link is "INTERNALLY CLOCKED", does this mean the T100P is providing clock, or does this mean the T100P is getting clock from the T1 line side (ergo getting clock from the Telco) ?? If you have sync = 0 then zttool

I'm considering using asterisk to replace an existing PBX in a 40 room hotel and would appreciate any comments, corrections or insight before I begin. Only 8 PSTN connections are initially required but since the guests need dial-up internet access in the rooms it has to be Frac-T1 as opposed to using FXO ports on a channel bank. IP phones are not an option strictly because of price. The

Hi all, Which is the practical (from your experience) limit of the number of X100P cards installed in a single Asterisk box? Asterisk can work reliable with 6 X100P cards in the same box? Anyone know when the 4 ports FXO Digium card will be available on the market? Many thanks, Dan P.S. Please do not aswer with RTFG ...tried before without success...:-))

Hi list! The stability of * with fax (or the lack of it) is causing me headaches. To solve it, I was thinking to put a TE205P card in the * box, connect the E1 pri on one port and a channelbank (I was thinking of the Rhino) on the other port in T1 mode. (Has anyone tried this??) The TE205P supports channel bridging which sounded like the ideal (but not cheap) solution when combined with a

Sorry for this. This is most likely a HOWTO or FAQ question, but it's so much information and documentation to wade through so I hope someone could take a minute to answer anyway. If not, no worries. I'll get to it sooner or later :) I'm trying to understand what Asterisk actually is and the basic workings... I think I've understand what I need to get going, except one thing.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've been seeing some strange issues recently as it relates to our Asterisk phone system. Let me give a little background on things: We have a T1 provided by Knology. We recently had an outage of Internet and phones. The following problems (except for the log messages) seemed to have cropped up after the outage. I was prompted by a friend of mine

Hi, I've read through the Asterisk handbook and I'd just like clarification from someone that's implemented the above before. Lets imagine I want to use the CallingCard application and don't want to tell a client to buy a channelbank (the analog extensions are too many to connect to FXS cards), I figure I could set them up as below: Original Existing Setup

Hi my question is: which is the best distribution to work with asterisk? thanks mark