similar to: Digium Wildcard TDM800P not working with DAHDI

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 .....

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 + 52 442 167 08 00 VoIP 523663899 d00d! cyberalph -------------- next part

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

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

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

The card you installed has FXO or FXS modules in it ????? are you getting your lines directly from the telco co??? Doug D On Mon 23/08/10 8:37 AM , Cassius Smith cassius at cassius.org sent: * -----Original Message----- * From: Todd Reese * Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion * To: asterisk-users at lists.digium.com [3] * Subject: [asterisk-users] Dahdi

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

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,

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

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

Sorry, I forgot to correct the message subject, so I am resending my doubt it below: Hi all! I wrote a very basic program in R, which has some loops ('for' and 'while'). Is there a way to print the 'for' (or while) indice while the program is running so that I am able to estimate where the program is and how long it will last? I mean, I want something like this:

Hi all, I have two questions 1 - I have the version 1.4.1 of R, and it doesn't have the 'agrep' function in the base library. Is there a way to make this funcion avaliable in R 1.4.1? I mean, how to 'copy' it from R 1.8.1 and 'paste' it in R 1.4.1? 2 - The AGREP function doesn't give me the Levenshtein distance (edit distance). Is there a function in R that does

Hello group, I am installing Asterisk on a new pc running Ubuntu 12.04.3 Server. ?I have dahdi-complete-2.7.0.1+2.7.0.1, and have tried 2.6.1 and 2.8.0-rc1 and 2.8.0-rc2. ?I am following along in the ASTERISK The Definitive Guide 4th Edition and on page 116 when I issue the command $ sudo /etc/init.d/dahdi start I get the following error: Loading DAHDI hardware modules: ? ?wctdm24xxp: done

"Marcos Sanches" <marcos.sanches at ipsos-opinion.com.br> wrote: Ls1<-length(s1) Ls2<-length(s2) for ( p in 1:ls1){ for (q in 1:ls2){ t1<-levenshteinFast(s1[p],s2[q]) ... Ls1=42000 Ls2=70000 I think I will wait for months untill this program ends. Do you have any sugestion to increase the speed? The first suggestion has to be "search HARD in the

Hi, We're running on a * 1.4.21 system. We run about 80 SIP Extensions, mainly ATCOM phones (and a few Snoms - 300 and 360), and have an additional 80 IAX2 extensions to iaxmodem devices for fax2email. We are rapidly growing and will be adding an additional PRI trunk and grow to about 150 SIP & IAX2 extensions towards the end of the year. We have two Digium Wildcard TDM800P cards (8 x

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

Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and Celeron), and last days when I call from one extension to another of the same PBX after I dial the number the rings sound after 20 seconds. In the CLI log, when I debug the AGI, I see always goes good until dialparties.agi, and after that there are 20 seconds without any log, and so the ring sound. I've read

Hello: We have a TDM2433E Digium Card (12 FXS, 12 FXO) and Asterisk 1.8.7.0 running. Everything seems to be ok but call transfers. This is the issue: *A, B, C and D are in FXS ports*. 1) A calls B. B anwers. 2) B tries to transfer the call to C dialing *2 (code for attended transfer). 3) A hears MOH. B dials number C. 4) Asterisk says the dialed number is incorrect or non existing. We tried

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

I changed my R console configurations, for example, the letters are white, the background is black, etc,... Then I saved this new configuration. What should I do if I want to have this new configuration everytime I open R? TIA