similar to: telco access ?s -- PRI, T1, POTS?

Greetings list, I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. If anyone's had experience using channel

I've been replacing an old Toshiba DK switch with an Asterisk solution. I'm needing a solution for fax machines that works as well as a POTS line from my carrier. If the POTS line is the solution, I'll keep it, but I'd rather move away from that. Here's what I'm thinking...will it work? I would use a dual-port Digium T1 card. In one port, I'd terminate a telco PRI

Hi guys, I've run into a problem that I can't figure out on a bunch of handsets I have running into a Rhino Equipment 24-port FXS channel bank hooked up to a T100P and running asterisk-0.9.0 and the associated stable Zaptel release. The sidetone (your own voice that you hear in your handset, built in for comfort) is noticeably louder than it should be, and it doesn't seem to

Hi all, I'm sure something similar has been discussed, but one can only wade through the archives for so long. I'm setting up a T1 and my telco has a bunch of questions it wants me to answer. I know much of the TE110p is configurable to do any of this, but I wanted to know if there is an optimal or preferred setup. Any help would be appreciated. Here is the quiz my telco is giving me:

Hi I've had an asterisk setup for the past 15 months, based on the debian asterisk packaging. Until late August of this year, I had no problems once initial setup was complete- the system worked essentially flawlessly. Since August I have been having exceedingly infuriating intermittent problems that are causing me occasional periods of nasty trouble: 1. No Dial Tone. Every Sunday night at

Hi all, I'm sure something similar has been discussed, but one can only wade through the archives for so long. I'm setting up a T1 and my telco has a bunch of questions it wants me to answer. I know much of the TE110p is configurable to do any of this, but I wanted to know if there is an optimal or preferred setup. Any help would be appreciated. Here is the quiz my telco is giving me:

Hi again, I'm trying to setup our non pri T1 (they call it a Long Distance T1), our current pbx has the signaling set to E&M, I can set em in zapata.conf, but I'm trying to track down the proper entries for the zaptel.conf file. The digium docs only show a PRI example. Our current system has these settings: Signalling: E&M Framing mode: ESF Line Coding: B8SZ here's my

Paul, Ah, I see. Our echo is largly under control now. It took me a while to figure out the gains and get them tuned, and now the echo only leaves very small artifacts. Nonetheless, this still provokes the odd complaint here and there. We use VOIP for outgoing calls when our POTS lines are congested, and we find zero echo during those calls. Therefore, I assume that our handsets (Cisco

Hello - I have a non-PRI T1 setup and have been making outgoing calls for several months no problem. I have Zapata.conf setup for fxs_ks on these channels. How do I take incoming calls on these same channels? Do I need to change the signaling? Any help is appreciated. Thank you, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL:

Ok, I'm about to take the plunge, and am trying to decide between Channelized T1 E&M and PRI. I'm getting an "Integrated T1" which will have data and voice capability, all plugged directly into my digium single T1 card. In either case the data piece looks pretty straighforward, just setup the channel properly, hand it off to the linux hdlc layer, and route away.... the

Hi, I have a client that has a 24 channel voice T1 that I have been using e&m signalling over for a number of years. The local telco finally got an ISDN switch and wants to move them to PRI. I didn't see this as a big problem - I've done a few others on this particular Caribbean island without issues, but this would be the first time with a Xorcom unit involved. We tried to do

Could anyone with any experience with * over a non-PRI T1 help this newbie? I have a fractional T1 that is working fine through a channel bank, but I can't get any response on * using a T400P. My analog line and the extensions work fine, but the T400P port 1 shows a red led (what exactly does that tell me?). I brought up zttool which shows a condition red also, but ztcfg shows nothing amiss.

We have 2 T1's coming from our phone switch to a digium TE220B. We have managed to get CPN and the extension outpulsed from the switch, but call setups are really slow. Our T1's are set up as E&M Wink, and they send us the last 5 digits dialed followed by the 10 digit calling party number (we couldn't get the switch to be happy with *CPN*+5* to use featd). We are using asterisk

Has anyone here used T1s with RBS with asterisk? Cary Fitch

I am trying to receive both CID and DNIS from the telco through a non-pri T1. Currently I have the T1 setup and operational both outbound and inbound calls are completed as should be expected. The calls came in and were placed in the context specified in zapata.conf on exten => s,1. I have requested that the telco provide callerid (they call it ANI) along with 10 digit dnis for my 800

I want to replace a custom PBX, that is infront on a IVR system based on OLD NMS AG-E1 Card. The Cards is configurated with CAS Digitalmode, someone can give me some info about Digim Cards CAS configuration i need a conversion Table? I wanto to don't touch configuration on winbox, i want only replace HWPBX box with asterisk. Diagram Telco E1 ===>Proprietary PBX========(CAS)===>IVR

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

Hello I need to find a recent and neutral comparison of the major products available to connect an Asterisk server to the telephone network, whether ISDN (BRI) or PSTN, and through a PCI card or some external box. I'm told there are less issues (echo, stability) with external boxes compared to PCI cards. Apparently, the main brands are Digium, Sangoma, Rhino Equipment, Patton, and

All, I have done research on VoIP for some time now. I'm a Cisco certified Network Engineer however Telecom is not my strongest suit. I've been a part of this mailing list for sometime but my delusions of grandeur in migrating our 25 year old phone system to a new platform have been on the back burner, until now. I have found my company is moving to a new location(building) and this

I am forklifting a Merridian option 51c with 112 Nortel Digital Handsets and 400 analog units. For the analog units I have quotes for 9 ADIT 600 48 port fxs units and 17 Rhino 24 port FXS channel banks. I have used neither. Which is the best choice? The price difference is not that great. I am looking at Citelinks 24 port Handset Gateway for the Nortel Digital units. (Any other suggestions