similar to: T1, FoneBRIDGE, and dropped D-Channel

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

I am looking for input on wether the 4 port T-1 foneBridge is a useful device in a asterisk deployment. Would it be better to loadup a extra asterisk server to trunk the T-1's via IAX or is TDMoE safe in production enviroments. We are only looking at 2 T-1's, one for pstn and the other to a channel bank. Any advice? Bruce Reeves Nortex Networks -------------- next part -------------- An

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 all, i am trying to setup my first t1 in asterisk, i have been using asterisk for several years but ahve never needed a t1 line before. I have a sangoma card already in the server with 4fxo ports. Now i ordered two single port t1 line cards from sangoma for the two servers i am connecting with the point 2 point t1. I am currently at the location trying to set things up but have some

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

Hi, I'm using heartbeat as a failover for my asterisk server. on the active server 1 i have 10.10.10.1 eth0 10.10.10.3 secondary eth0 asterisk listens to the secondary ip, so that if server 1 fails, server 2 will then get that IP. so if server 1 fails, server 2 will have the IP 10.10.10.2 eth0 10.10.10.3 secondary eth0 problem is i have to bind asterisk to the secondary IP if dont, i

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'll keep asking weird questions here: I have two T1 WAN circuits originating at my asterisk box and terminating at Asterisk servers at remote sites. Currently I use zaptel ppp assigned to all 24 channels for data. Now I would like to split off a few of these channels for voice trunks (I should use VOIP, but at the moment I do not have fast enough servers for the codecs) At one site

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

> Close. 10/100mbps Ethernet uses wires 1,2,3,6 but that is pair 2 & 3. > Pair > one is the pair up the dead center (pins 4&5), pair 2 is pins 1&2, pair 3 > is > 3&6 and pair 4 is 7&8. A T1 uses pairs 1&2, which is why you can't use a > regular crossover cable for a T1 crossover, but you can use a regular > ethernet patch cable as a T1 patch

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

I am connecting a t100p to a b8zs, superframe, loopstart t1. Previously I've attached to e&m wink and pri lines with no problems; however I seem to be missing something. Should it be fxols in the zaptel.conf (smartjack to x100p) or fxsls? With e&m wink the dnis set was 100, so it was easy to make an extension 100; and it would answer incoming calls. How do you get a loopstart to

Hi, i am having a problem with my point to point t1, which is being resolved and is a seperate issue. sangoma support has been a huge help and i am waiting on verizon to increase the signal output of the smartjack. But my issue is that in the meantime my fallover extensions arent working. Well they are on the CPE side but not on the NET side. The NET side still thinks its making calls, they

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 everyone, I'm using redfone fonebridge to have Pstn connectivity on my asterisk box. I can receive in coming calls however outgoing calls don't go to provider. It's seems it's a span config problem. Because in systemconf when I try to config span as follow span=1,0,2,ccs,hdb3,crc4 And start dhadi I have the following message Unable to

SYNOPSIS Erratic red alarm T1 PRI on asterisk, but zttool running concurrently during alarm shows no errors, irq misses, or alarms, on any span. Using asterisk and quad Digium T405P, configured as follows: Span 1 connects to ISDN PRI (fractional 8 B channels, D channel 24). Span 2 connects to T1 Mux and analog stations. Span 3 connects to ISDN PRI Nortel BCM hybrid key system digital trunk.

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