similar to: Hardware question about Redfone's foneBridge

We have been running around than 40 asterisk servers running on Debian Squeeze for last three years, handling traffic of more than few hundred thousand calls per day. Our setup's PRI-banks were using Redfone's Fonebridge. We had PRIs from multiple telephony providers. And Redfone's Fonebridge handled all that easily. But all good things come to end. Redfone's Fonebridge was not

So here is my setup. Hardware: Intel P3 1.2 Ghz 1 GB RAM 36 GB Drives Mirrored Software: CentOS 5 2.6.18 Kernel Asterisk 1.4.14 Zaptel 1.4.7 (redfone) LIbpri 1.4.2 I'm using TDMoE with my PRI using a product called fonebridge from a company called redfone. They require that I use their own build of zaptel and I am trying to figure out if the problem is with them or something else. The

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

Has anyone every attempted to set up a mini PC to achieve much the same functionality as the fonebridge box? The sort of thing I'm imagining is a micro itx board & case in a completely solid state configuration (flash disk, maybe a psu fan but only if really required), with a TE210P (or equiv) card(s). The sole purpose of this would be to be a bridge between E1 lines and TDMoE. The

Hi, Some help with dimensioning the server will be gladly accepted. -50 sip phones (g729) or g711(to avoid transcoding) in LAN -an asterisk server (1.4) doing normal pbx functions + voicemail in the same LAN -Some sporadic conferencing with no more than 2 sip phones and maybe 2 or 3 calls coming from the E1 for a total of 5 people in a conference. The asterisk server will get an E1(pri) via one

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

Has anyone on the list used a Soekris engineering PC as a TDM - Ethernet bridge? For example something like a net4801 with a TE110p in it and then using TDMoE to get it into a bigger server where the call processing proper will occur. Anyone know if it might handle a quadspan card ok? (no transcoding, just pure PRI to TDMoE bridging). Craig

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

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:

I've been evaluating Asterisk for a while, and things seem to be going very well. The issue of redundancy and automatic fail-over is now on my mind. I searched the archives and googled for solutions, but didn't really come up with much. We'll be using queues (modified), which precludes some of the standard redundancy solutions, since the queue needs to know all the agents

First, Something seems to be wrong with the list. I'm not the only person who has expressed seeing their messages either arrive late, or not at all. With that out of the way.. Is anyone aware of any type of failover device for PRI on asterisk? I've found the ISDNGuard, however it is currently not made in the U.S., nor does it run on U.S. power. Is anyone aware of a device that will

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

Hi List, I am experiencing an issue in a server that I have installed asterisk; configured an AVM FRITZ card to work with the capi module. Once istalled the card works perfect; however every time I reboot the machine I found that I have to re install the capi4k-utils before I can load asterisk otherwise the capi module will not loadup. Can anyone direct me in the right direction in order to

RAD appear to have bucketloads of products which bridge between various interfaces (E1, BRI, POTS) and their own TDMoIP protocol. The attractive thing about them for me is their availability in Australia. The voip wiki says not much about it (http://www.voip-info.org/wiki/view/TDMoIP), and certainly nothing about if there is any way to get Asterisk to talk TDMoIP. Despite the name, TDMoIP

What are the options to hook a T1 card up to a laptop running * ? Are there USB or PCMCIA T1 cards ? Has anyone tried a USB to PCI adapter as such : http://www.mobl.com/expansion/products/cardbus_expansion/1slot/ It looks nice but cost 1k how about the linux drivers ? the goal is to have a portable demo system thanks - Arnaud

Hello Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. Thank you.

Was'nt sure if this mail got through earlier: I have been having a weird issue with my telco's ISDN PRI occasionally resetting on a incoming call, i suspect it to possibly be a timing issue since some of the incoming call work. This problem happens very frequently. I am using asterisk-1.6.0.1 with libpri-1.4.9, the asterisk server is connected viw TDMoE to a Redfone Fonebridge into

I am preparing to connect an asterisk box with a redfone fonebridge to a T1 service provider. I am doing this by testing first with another asterisk and a Sangoma card playing the role of telco. I formerly had this test configuration operating flawlessly as a PRI connection. But I discovered that I will need to use E&M, thus I've chosen the parameters as described in the subject

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

Hi Everybody, I'm trying to connect an asterisk box to a provider using Redfone Fonebridge dual E1. Installation seems to run correctly only i can't get the D Channel up and i have the following error displayed. DYN/ SPAN ethmf <mac_address_fo_fbport> Expected seq no 0 , but received 3456 instead . This error keep scrolling until i disconnect the