search for: tdm800p

Hi all I sold new TDM800P card with 8 FXO ports, someone know if can be use this card on AsteriskNOW or trixbox? What can i do for use this card? Thanks. ---------- RafaelCanchola Product Development Engineer, FonetGlobal Inc. rcm at fonetglobal.com http://www.fonetglobal.com Ph. + 52 800 022 10 21 ext. 214 + 52...

Dear all I have asterisk 1.4.11 on Cent 4.3 i have faceing some problem i have TDM800P 8 port FXO card when i terminate PSTN line on this port can make outgoing call it is working fine but incomming call not handling ...when i call from outside to this line it is rinning but no one call land on my asterisk no debug in asterisk some time it land but most of time not ..... ----PGP...

Hi All, I have a server running Fedora 14, kernel 2.6.31.14, Asterisk 1.6.2.17.2 and Dahdi 2.4.1. I have the wctdm24xxp+ loaded with a Wildcard TDM800P with 8 FXO ports When a call is placed extension to extension, there is no problem... When an extension is used to dial out, the called number answers, but the server only detects the answer about 3 seconds later .. The outgoing line is TDM on port 2 of the TDM800P. The same hardware...

Hello everyone, How can I let Digium Wildcard TDM800P work successfully with DAHDI? Because the Centos recognizes the card but I can't get the analog card working with DAHDI. Thanks in advance, Gilberto -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments...

I'm new to Asterisk and inherited this project so I apologize if this question has been asked a hundred time before. I did start with Google but I may not be asking the right questions, because I wasn't finding any answers. I have Asterisk 1.4.24 and FreePBX 2.5 running and using a Digium TDM800P to interface with our six analog phone lines from the telco. Currently I have a single trunk setup ZAP/G0 that will allow all incoming call go to your IVR and allow outgoing calls. My problem is I need two of those lines to be routed to one IVR for our tech support people and the other four to be...

I just installed a TE820 octal span T1 card, and it's not showing up in dahdi_hardware output. This was installed into a test machine that already has a TDM800P card in it, and that one is showing up and working fine. Is there some kernel module that I'm missing? Lspci: 05:04.0 Ethernet controller: Digium, Inc. Wildcard TDM800P 8-port analog card (rev 11) 21:08.0 Communication controller: Digium, Inc. Device 1820 (rev 02) Lsdahdi: ### Span 1: WCTD...

...ribanks/drivers/xppdrv/sync' I found some steps at?https://issues.asterisk.org/jira/browse/DAHLIN-325?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel Here is the output of the commands listed there: $ sudo lspci -d d161:* -v 04:01.0 Ethernet controller: Digium, Inc. Wildcard TDM800P 8-port analog card (rev 11) ? ? ? ? Subsystem: Digium, Inc. Wildcard TDM800P 8-port analog card ? ? ? ? Flags: bus master, medium devsel, latency 32, IRQ 16 ? ? ? ? I/O ports at d000 [size=256] ? ? ? ? Memory at f7c20000 (32-bit, non-prefetchable) [size=1K] ? ? ? ? Expansion ROM at f7c00000 [disabl...

...From: Todd Reese * Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion * To: asterisk-users at lists.digium.com [3] * Subject: [asterisk-users] Dahdi install gone wrong * Date: Mon, 23 Aug 2010 10:26:58 -0400 * * Hi All, * * * I've got a project installing a Digium TDM800P card with 8 FXO's in an * Asterisk box. * * * The computer is running Slackware 13.1 and I've installed the current * Dahdi and Asterisk 1.6.2.11. * * * I've installed several boxes that are pure VOIP but, I haven't installed * a Dahdi interface and I'm stumpe...

I recently read about the following new technologies from Digium. Has anyone tried the new HPEC or knows when it will be available? TDM800P and HPEC The TDM800P is an 8-port analog telephony interface card, so it fills the gap between Digium's 4-port and 24-port cards. Analog phones and POTS lines are going to be with us for some time, and demand for support for them remains high. The TDM800P is a bus-mastered PCI card, which me...

Hi there dear users and dear developers of Asterisk! I've got a maybe strange idea, let's see if you laugh or think it's reasonable J I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering machine). Each analog line can be reached through a phonenumber, since they are each connected to my telephone provider. Yes expensive I know J The challenge: I'd like to somehow verify that my 1) TDM800P cards and 2) the analog lines, and 3) my operator...

...re maintenance releases that fix various known issues. See the ChangeLog included in the releases for a full list of changes. The ChangeLogs are also available separately on the ftp site. This release also contains support for Digium's new 32 channel hardware echo canceler (VPMADT032) for the TDM800P and TDM2400P. Warning to TDM800P and TDM2400P users with FXO modules: Unless your TDM card contains a VPM100M echo canceler, you will notice an increase in your volume levels after upgrading to this release of Zaptel. You may wish to compensate for this change in zapata.conf. Both releases are a...

...re maintenance releases that fix various known issues. See the ChangeLog included in the releases for a full list of changes. The ChangeLogs are also available separately on the ftp site. This release also contains support for Digium's new 32 channel hardware echo canceler (VPMADT032) for the TDM800P and TDM2400P. Warning to TDM800P and TDM2400P users with FXO modules: Unless your TDM card contains a VPM100M echo canceler, you will notice an increase in your volume levels after upgrading to this release of Zaptel. You may wish to compensate for this change in zapata.conf. Both releases are a...

Hi All, This is a request for testing for users of Asterisk 1.4 or 1.6 with any of the following Digium VoiceBus based cards: TDM2400P, AEX2400, AEX800, TDM800P, TDM410, TE120P, TE121, and/or TE122. From a practical standpoint, this branch should allow these boards to work in more systems / configurations where IRQ misses caused problems in the past. The updated wctdm24xxp and wcte12xp drivers are based on Zaptel 1.4 and work by dynamically increasin...

Hi, I'm calling asterisk with a swedish PSTN-phone line with caller presentation (DTMF) activated. I'm using asterisk 1.4.20.1 and cannot upgrade unfortunately, so I have to stay with this release. I use a TDM800P 8 channel PSTN card working as answering phones (I connect a phoneline with carrier signal to my TDM-card). Using zaptel-1.4.12.1. I verified that the DTMF tones of the number is really sent on the PSTN by connecting a number presentation box between. I see the incoming number when I dial into ast...

We have several offices running Asterisk version 1.4.20.1, and OSLEC with Rhino R4FXO-EC and one running a Digium TDM800P card for interface to analog lines. All offices are running Snom 300 phones. Phones all have static addresses and are on the same physical network as the server. The problem we are having is that every so often we get someone calling in where we can hear their voice, but they can't hear u...

Hi I have a server with the last version of asterisk branches, zaptel branches, 2 Digium Card with TDM800P 16 HPEC channels (Echo Cancelation), 40 Grandstream BT200 and 10 Grandstream GXP2000. zapata.conf echocancel=64 rxgain=0 txgain=0 when i place a call o receive a call, I finish a sentence i hear a ssssssss, AND when the both side talks at the same time i have choppy audio. Any help i...

Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo cancelation and a quad FXO card. We have 4 analog lines, one of which is a Cellphone line for least cost routing. The problem I am having is dialing out using DTMF signalling. At the moment I am making do with Pulse dialing through the 3 analog lines. I can recieve ca...

What is involved in provisioning Asterisk to use a multiline analog service from our local telco? I will only have one twisted pair entering in on a OpenVox card but am not sure how Asterisk interprets and deals with two incoming calls and/or two outgoing calls? Thanks! jlc

I have recently moved an asterisk system to a new location. This location is experiencing terrible echo. I installed the HPEC from Digium but that has caused a new problem. When HPEC is enabled and more that 16 taps are used, the audio from the outside caller gets clipped. Instead of hearing: Hello, my name is Mike one hears He o, m ame ike If the taps are set to less than 16, there

Hi, I am a newbie to Asterisk; need help understanding three-way conferencing & call-transfer features implemented over standard extensions i.e. on a TDM800P card (4 FXO + 4FXS) In Asterisk I have observed that if an extension is already participating in an active call (e.g. Ext A & Ext B communicating): 1. An incoming call to one of these active extensions would be presented with call-wait beeps (e.g. Ext A receives call-wait beeps as Ext C is...