similar to: Configuring new system for a non-profitorganization

Hi, The PBX at the Colorado Organization for Victims' Assistance fried as a result of the building power being cycled. I'm now in the process of building an * system to replace the failed PBX. Minimum cost is the priority. I have a T100P card installed in the new system, and I am about to order integrated T1 services from the "CBeyond" company. They will require eight

I don't know if anyone had used these boxes yet. I've installed some, but not connecting to an * system. The guys who ordered them said they ran about $600. They are pretty cool in that you designate what the port will be either Data, FXS, or FXO(not fully implemented yet) This is a "NEW" price too, not a refurbed price. I don't know about anyone else, but I've not

I'm having some trouble with placing some iax calls over an openvpn: Setup A is a 1.8GHz Celeron, T100P attached to a Zhone Zplex. Setup B is a 266MHz P2, T100P attached to a Zhone Zplex. Setup C is a 700MHz P3, T100P attached to an Adtran TA 750. Setup D is a 233MHz Pentium, with an X100P. Setups A and B are on the same physical network. IAX calls routed between them work fine. Setup D is

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

I'm experiencing a very strange problem with SIP calls with a CLEC (CBeyond). The downstream audio with the telephone on mute is excellent. However, when there is upstream audio (even breathing) from the mic, the downstream audio is clipped and sometimes dropped. The strange thing is, if I Monitor the call, the downstream audio in the wav file is perfect, even though there was clipping

I'm experiencing a very strange problem with SIP calls with a CLEC (CBeyond). The downstream audio with the telephone on mute is excellent. However, when there is upstream audio (i.e., breathing) from the mic, the downstream audio is clipped and sometimes dropped. The strange thing is, if I Monitor the call, the downstream audio in the wav file is perfect, even though there was clipping

An example for Radius is calling cards.. I can use * for this kind of service... With platforms that use Radius Server. -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Lunes, 17 de Noviembre de 2003 07:16 p.m. Para: asterisk-users@lists.digium.com Asunto:

Hi! Is there anyone that are using a E1-channelbank and have any tips about some type? Im looking at the TE410P and use one port for a PRI (Euro-ISDN, I think we're using some slightly modified version here in Sweden, but I'll check that tomorrow) and connect one port to a channelbank for 30 analogue telephones. It would also be great to get callerid on the analogue phones, so it would

I've got an * system that is having some real problems with 1.0.2. The biggest problem is that calls going through my T100P get choppy for about 10 seconds every 1 or 2 minutes. Asterisk is running on a debian stable system with current packages. The T100P is plugged into a Adit Channelbank with 8 POTS lines hooked up to the Channelbank. I've watched the vritual memory and CPU status on

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

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

I would recommend that you look at the Pika Technologies Daytona MM board. It has onboard DSP and onboard analog bridging taking up much less horsepower. Please contact me off-list if you would like more information. Bill Hunt Stroudwater Contact Point 207 347 8080 x219 877 870 1234 Toll Free www.stroudwater.com "Realize the Value of Customer Contact!"TM This e-mail is intended

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

Hi, Does anyone know whether faxes work good when they're connectected to a channelbank ? regards, Leon

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

Merry Christmas from the Colorado Organization for Victims' Assistance. Our (Comdial) PBX fried after a power failure. Thanks to Mark Spencer, Digium, VCCH, and the friends who support this group, we are now back "on the air". We wish everyone good health for the coming year.

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

In our experience, the total cost of softphones(money, reduced sound quality and lower reliability) in a large call center environment is actually greater over time than the cost of a channelbank and cheap analog headphones. We've tried 2 softphones, 2 kinds of SIP VOIP hardphones, 2 kinds of SIP analog adapters and we've tried channelbanks over the last 3 years. Right now we are half done

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

Has anyone had any experience connecting Asterisk to Cbeyond's SIPconnect service (http://www.sipconnect.info)? Any opinions? Thanks, -Brian