similar to: Gateway E1 <=> Asterisk ?

Does any have experience with E1 telephony support plus asterisk in costa rica ? Regards, Luis Morales -- --------------------------------------------------------------------------------- Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 --------------------------------------------------------------------------------- "Empieza por hacer lo necesario, luego lo que es posible... y

I'm looking for recommendations for standalond PRI to SIP converters. (Needs to be outside the asterisk box - so a PCIe card won't do) I've used redfone but this project doesn't need the redundancy features... Thanks!

I'm currently looking to interconnect my Asterisk PBX system with the PSTN via a digital PRI/T1. I know a multitude of options exist for internal PCI cards (Digium/Sangoma/Rhino), I was wondering if anyone has any experience or recommendations of external PRI media gateways that support SIP. So far I've found: VegaStream Vega 400 Audiocodes Mediant 2000 MediaTrix 1531 However they are

Does anyone know of a hardware adapter that can take ISDN BRI frames (I.430) and encapsulate them in Ethernet (any form, but TDMoE would be really cool), in much the same way that the redfone does for PRI? (yes I have asked this before in looser terms, but it was a while ago :) Would anyone find such a device useful? Thanks James

Hello, I am trying to Bridge 2 E1 interfaces over a long distance link exactly the same way Redfone does. How can asterisk be configured to do that? Best regards Arinze Izukanne ___________________________________________________________ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good http://uk.promotions.yahoo.com/forgood/environment.html -------------- next

Hi i have buy a Audiocode Median 2000 VoIP Gateway and connect it on : 1 E1 30 channels 1 Lan Port Anyone use this equipements with asterisk ? because i am search a config sample for AudioCode and for Asterisk (i am new in VoIP). I want that all calls arrives on the AudioCode are sent to the asterisk by SIP (trunk ?) and all outgoing call from Asterisk are sent to the AudioCode. 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

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

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:

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 List, I'm finding a solution to provide failover and load balancing features to my IVR system. Anyone suggest me what is the best solution please?. what the hardware I should use ?. I heard about RedFone, but someone on the mail list said that it is not good because TDMoE module in asterisk is not so stable and TDMoE is stale. And It seems that RedFone doesn't not support load

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 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

Hello, I am using Queue application with multiple agents in each queue. I want to set the CDR(userfield) for each cdr based on the agent answering the call. Is it possible to do this? Thanks

Does anyone have any information for me? Welinghton. Citando Welinghton Magno Guimaraes <welinghton.guimaraes at ufvjm.edu.br>: > Hello! > ? > I am setting up an Asterisk server with a Mediant 1000 (Audiocodes) > to make external links. Does anyone have any manual or instructions on > how to proceed? > ? > Asterisk ?=>? Mediant 1000 (AudioCodes) ?=>?

Does anyone have any experience using a PRI gateway, I am looking for a way to have multiple asterisk boxes use one PRI, and send that over the network. I herd there are copper gateway devices (like a X100P card, only it registers with asterisk using sip, and it doesn't have to be physically connected to the box) Does anyone have any experience with a PRI gateway? And could tell me the cost

Hello Group, I have been going through all the chit-chat about TTS and the various engines available to integrate with Asterisk incl. flite/festival, espeak, Nuance etc but I am wondering if anyone's tried any or all of these to compile on a Sparc based Solaris platform? If not, then what is the best way for me to accomplish a production environment TTS service when most of my servers or the

I know I asked this some time back, and I got no response then, and neither did someone who asked at the start of 2009 either by the looks of it (other than a reply from me to use a PCI card!!!) However I now have a client who'd bought one of these boxes and installed ISDN2e - against my advice (because I got no feedback here, therefore they're junk with asterisk, right?) However

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

Hi it's possible into Asterisk 1.6.x to limit a call at 120 mn ? after 120mn, hangup and the customer call a new time thanks olivier