similar to: foneBRIDGE2 setup

Hello, I'm trying to decide between the foneBRIDGE2 ($1135) and foneBRIDGE2-EC ($1610). Has anyone here directly compared the two? Would we really suffer without the onboard echo cancellation? The manufacturer's site doesn't really give much helpful information about choosing one over the other. Thanks. -- Kevin DeGraaf

I've been asked to look at deploying Asterisk in a high availability environment and I've been looking so I've been searching for methods to decouple the voice PRI circuits from the Asterisk server so failover to another server could take place. I've been looking at the RedFone foneBRIDGE2 2e1 product here:

I'm looking for reports of recent experience with redfone fonebridge2 (with echo can) TDMoE gizmos. Anybody? Good? Bad? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080609/23aeddd6/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s

We have been using Red-fone foneBridge2 Quad T1/E1 for last few years. As these devices are not available anymore, we are looking for alternatives. Are there any similar devices available ? -- Regards, Tirveni Yadav www.udyansh.org What is this Universe ? From what it arises ? Into what does it go? In freedom it arises, In freedom it rests and into freedom it melts away. Upanishads.

Hi I've recently released the daemontools scripts I use to run both Asterisk and Flash Operator Panel, and a macro to tell whether today is a holiday or not and jump to different dialplan places accordingly. They are here: daemontools scripts: http://www.bisente.com/blog/2007/04/27/spanish-asterisk-y-daemontools-spanishenglish-asterisk-and-daemontools-english/?lan=english is-holiday macro:

Hello, Can someone outline what tutorials will be covered at this year's AstiCon in AZ? Are the tutorials going to be worthwhile for fellow Asterisk users/admins that have been actively building, running, and administering Asterisk boxes for years? I have a suggestion for one demo that would certainly interest me and other more advanced users. I would *love* to see a hands on live

I'm having (I think) timing issues in relation to bridged T1-T1 calls via dynamic spans. Fax calls are intermittently working, but voice is fine. My box has a Sangoma A400 inside it as the primary Zaptel timing source. My T1 PRIs that are hooked to the box come in via a foneBRIDGE2 (dynamic TDMoE spans). PRI #1 is the telco and PRI #2 is an existing Comdial FX-II. For some reason, bridged TDM

We tried them out early last year when we were looking at a large deployment and they gave us a lot of the redundancy that we wanted. However we did run into issues where calls seemed to get caught up in the system. It was as far as we could tell rather random. No consistency to it at all. Asterisk hung up the call but the telco side of the line didn't actually hang up. The channel was left

For high-availability Asterisk when using PRI Lines, you might also want to check out our product (FSV-4PFS). It's available at www.failsafevoip.com Bill Also to add to the last post does this device have hardware echo cancelation? if it does it could be a great replacement, if not may not be what I'd really want to use. Thanks, tom I've been looking at the RedFone foneBRIDGE2

I'm scoping out HA for a relatively simple Office/Call Center PBX. Current setup uses a TE412P with 4 PRI our telco with SIP hard/soft phones for users. Some outbound also goes to a SIP provider. Active/Active looks to be too much hassle for an installation this size, so we're looking at adding an extra * in an active/passive configuration with Linux-HA in between them. Does anyone

Need to add some fxs and fxo ports behind a fonebridge2 box any recommendations a channel bank Thanks Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070829/1131cad4/attachment.htm

Hey guys, I am trying to diagnose a hum in my FXO lines I am using an Adtran 750 with 8 FXO ports. I am getting a pretty bad hum on the line during a call. I have checked the Telco side of the 66 block and there is no hum there so it's my problem to fix. I have tried to lower the gain but that reduces the call volume to much. Where else should I be looking? Setup as follows Dell

Hello List Does anybody have any experience with the FoneBridge line of products from RedFone? I think their HA implementation sounds interesting, and like the prospect of having dedicated hardware for our PRI connections. Kind Regards Jon Leren Sch?pzinsky

I am trying to sort out a configuration problem with y slave servers, more on that in a following message, but while I was examining the system log on one of those servers I saw this: Nov 3 12:30:43 inet04 named[18110]: client 174.88.12.107#60136: query (cache) 'www.soundunwound.com/A/IN' denied Nov 3 12:30:43 inet04 named[18110]: client 174.88.12.107#60140: query (cache)

Ok Guys, I've done a tonne of hunting around on this problem, but can't find much help. I'm running: asterisk 1.4.19.1 libpri 1.4.3 and zaptel 1.4.9.2 which I believe has been modified by RedFone to add the ztd-ethmf module. My interface is a RedFone foneBridge2 4 Span; and I'm connecting to a BT E1 PRI / ISDN30e with 15 lines on span 1, and a legacy Panasonic PBX on span 4. Upon

Hello, I want to create channel bank in this case: "channel bank" |-----------------------------------------| | FXS,FXO<----->TDMoE<--|---------------------------------->Asterisk |-----------------------------------------| How can it?

Hello list, AstPligg is a new Digg-like website devoted to * and VoIP news. At the moment, it's in beta stage and very basic - no fancy custom templates. It allows posting new stories, comments on stories, RSS feeds and tags. Still, it can be very useful, as the number of * sites and blogs grows every day, and keeping track of what is hot in the * world is increasingly difficult. Yes,

Here you go folks: ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf If someone would be so kind as to upload to the wiki, it will be much appriciated. Thank you all who replied to my poll questions. As always, I hope this help. JR -- JR Richardson Engineering for the Masses

Hello! I have problems with audio in conference zap sip, I have choppy audio. I believe this problem is cause by de echo canceller from the fonebridge that I use in my system. Can someone explain me how I can disable the echo canceller form the fonebridge? I'm using dual port T1/E1 foneBRIDGE2 Thanks!! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Is there any mailing list specifically for zaptel-users ? If yes, then, sorry to bother you people, i could not locate it. Can anybody post a link to it ? If not, then i hope, this is the right place to ask zaptel related questions. I'm using vanilla zaptel 1.4.3 on custom Linux 2.6.15.1. I've have a Redfone FoneBridge2 (TDMoE) gateway which i want to use with the zap drivers. It