similar to: [BULK] Fonebridge2

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'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

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

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 I've published my Asterisk/foneBRIDGE2/heartbeat setup: config files, scripts... along with a brief description of the architecture and working of the cluster. It's available here: http://www.bisente.com/blog/2007/08/26/asterisk-cluster-fonebridge2/? lan=english Hope somebody finds it useful. :) Regards -- Vicente Aguilar <bisente at bisente.com> |

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 What soundfile format, is the one that uses least transcoding during playback? As I can see, I can choose wav or gsm. What sucks least cpu power, during playback to example a Zap channel? I would guess wav, but is this correct? Kind Regards Jon Leren Sch?pzinsky -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.11/496 -

I hope someone can help me out with this issue. It has been dogging me for months and I can't seem to get it to go away. I have a Rhino Ceros box running Asterisk 1.4.21.2 connected via eth0 (nVidia MCP61 Ethernet) to a RedFone FoneBRIDGE2 dual-port with EC. The FB is the latest hardware rev and the latest firmware. I'm running the latest fonulator version and I'm running Zap-1.4.11

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.

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'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:

Hello List. I just got my new PS3 yesterday, and first thing I did was of course to install Linux, and then compile asterisk, and it worked without any problems. My question is this... Is anybody looking into using the Cell processor for G729 enc/dec? Using the 6 SPE processing units available, you should be able to enc/dec a whole lot of channels at one time. Looking at the

Asterisk Project Security Advisory - AST-2008-012 +------------------------------------------------------------------------+ | Product | Asterisk | |----------------------+-------------------------------------------------| | Summary | Remote crash vulnerability in IAX2 |

Hello I am currently having a problem, that threatens to drive me insane... I cannot understand how Asterisk matches up a sip request with a peer. Here is my example: INVITE sip:87654321@192.168.100.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.59:5060;branch=z9hG4bK00d97b99;rport From: "1088200336" <sip:1088200336@192.168.100.59>;tag=as54af3e4d To: <sip:87654321@192.168.100.4>

Hello List Just want to check if anybody else is having this problem. Every time the PRI connections are disconnected, the card freezes, and I have to reload the driver, to make it work again. We are very seriously considering switching to Sangoma at this moment, due to this and other problems, but I want to know if there is a solution, and to make sure it isn't asterisk that's freezing

Asterisk Project Security Advisory - AST-2008-012 +------------------------------------------------------------------------+ | Product | Asterisk | |----------------------+-------------------------------------------------| | Summary | Remote crash vulnerability in IAX2 |

Hello List. We are having some problems using t.38 together with a Cisco voice router at one of our providers end. We are using the new digium asterisk fax module to generate the fax, and when we use together with our internal Audiocodes Mediant 2000 gateways, we have no issues what so ever, and the faxes go right through. When we send faxes to our other provider, who has cisco hardware

Hello List I am having a rather big problem with a sangoma A104 card, I just installed to replace a Digium TE410 card, that was acting up. But now we have a problem with the sangoma card. It runs great after being started, and calls proceed as normal, but after about 1 hour, it stops being able to make and receive calls. If I run wanpipemon debug, can see that the card still receives

Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I just dialed. Is this possible? So, if extension 4023 is John Doe, and I dial 4023, my display should read John Doe and not 4023. I am using a Polycom 501 by the way in