similar to: New user needs a hand starting

Hi, I have installed a TDM400P in a PC which runs RedHat Linux 3.2.2-5. There is an FXS module connected to the first slot of TDM400P and an FXO module connected to the fourth slot. This card is detected by the operationg system, which I was able to make sure using the output of lspci command. I modified the configuration files zaptel.conf, zapata.conf and extensions.conf. When I execute the

Here is my system: POWER SUPPLY 450W ALLIED ATX450P4 R(41) MB NFORCE2 A7N8X DELUXE ASUS RTL(Standard) CPU AMD|2500/333 ATHLON XP BARTON R(Standard) DDRAM 256M|DDR333 PC-2700 -K %(Standard) HD 40GB|WD 7200RPM 8MB WD400JB%(70) VGA ASUS|V8170MAGICII/T 64M MX440SE(58) CD ROM 56X|AOPEN CD-956 RTL(22) Besides the above parts, i have 2 X100P FXO PCI cards, and 1 TDM40B I bought 2 phones from radio

Hi all, I was wondering if anyone had any pointers on how to determine whether or not a module has gone wonky on the TDM400P? I have a 2 FXO (channels 3 and 4) and 2 FXS unit (channels 1 and 2). The bad (?) module in question is the FXO module on channel 3. I can't dial in to or out of that channel; dialing in gives a busy signal, dialing out just shows * hanging around after attempting a

Hi, I have a TDM400P running with only one FXO port running on a VIA EPIA MS10000 (1Ghz Via Eden). When I run zttest from Zaptel 1.2.1 it hang and when I interrupt it with Ctrl-C that is the result: ?anyone have some idea about why isn't working? Some additional info: # /usr/src/zaptel/zttest -v Opened pseudo zap interface, measuring accuracy... --- Results after 0 passes --- Best:

Hello, After installing on Ubuntu 10.04 using the tutorial on http://hmontoliu.blogspot.com/2010/03/installing-asterisk-and-freepbx-on.html I have a running instance of Asterisk. PROBLEM: result of dahdi_cfg: DAHDI Tools Version - 2.2.1 DAHDI Version: 2.2.1 Echo Canceller(s): MG2 Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2)

I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows the following: trixbox1*CLI> zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11*

I have two grandstream budtetone-100 and cisco 7960g phones. When I talk via speaker phone on either of the phones I get a lot of echo. Any suggestions? Also how do I turn on the mark echo canceller. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, April 13,

Hello, I have an issue with Digium TDM 400 card series. When I try to make outgoing call (PSTN call) for example, the Zap channel could not be created and busy channel message appeared. Below is the full log : [Feb 5 09:26:17] VERBOSE[3047] logger.c: -- Executing [s at macro- dialout-trunk:20] Dial("SIP/213-09648720", "ZAP/g1/08170709XXX|300|") in new stack [Feb

Hi, I have just started to setup and configure my asterisk box and am having trouble with it. I have a dlink nic in the box as well as the digium card. When the nic is in there by itself it works, but when I put the tdm400p in, there seems to be some sort of conflict with the network card. The tdm appears as another type of network card, which I don't think it should, should it? It shows

FC2, Asterisk 1.0.0, Zaptel 1.0.0 TDM400P Port 1 FXS Port 4 FXO Standard analogue handset plugged in with pstn line. Problem: I have 2 analog phones that I use, when plugged directly into pstn line both phones work perfectly, dialing no issues. When I plug the handsets into the TDM400P, one works perfectly the other drops random numbers. Its like the tone is slightly different on the second

All the 'unknown device' means is that your 'lspci' doesn't know what the card is. That's all. Nothing more. --Rob ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Aryanto Rachmad Sent: Tuesday, 4 October 2005 7:43 AM To: asterisk-users@lists.digium.com Subject:

Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the

Hi, I asked yesterday about a problem with x306 and IRQ sharing, didnt get much info, now, i was playing with lspci, and see something strange, lspci -v shows me the TDM400P card is on IRQ 7, and the SCSI card is also on IRQ 7, lspci -bv (from the man - b - shows "bus-centric view, as seen by the BUS and not by the kernel) shows me the TDM400P is on IRQ 5, why does the kernel puts it on

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

Sorry to post such a newb set of questions but I have been hammering about trying to get Asterisk running on FC2 machine reading everything available (I think that is what stuffed me, shouldn't have read it all :-) ). Config FC2 running Asterisk 1.0.0, with the h323 compiled in and installed correctly. Amazingly enough I have everything compiled correctly and installed. I am running a

Hello everybody, I have been googling for hours and also searched on http://www.voip-info.org/wiki-Asterisk, but I still can not find any information for the problem I have. So I hope one of you could help me out. I have actually very little experience in Asterisk and also Linux. But by following installation guide, luckily I could get asterisk working. That is only with SIP and IAX channels

Hi to all I'm a new user of TDM400P card. The configuration is OK and I have no problem for incoming call. When I try to place a outgoing call towards a PSTN number the call is not always answered. In other words TDM400P seems to not understand that the call has been accepted from the remote party. In other cases (different extension) the call is accepted succesfully. In my opinion TDM400P DSP

Hi I know this gets covered fairly regularly, but I've had a search through the archives and can't find anything dealing with this specifically - apologies if I've missed it though. I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN line, loading wcfxs with OPERMODE=UK. All's working well, except if I get an incoming call through my bt line, and the

Hi I was using the delivered Ubuntu 1.6.x packages but I wanted to look at gtalk integration so I downloaded, compiled and installed the source (after removing the Ubuntu packages) have installed the following: asterisk-1.8.0 dahdi-linux-complete-2.4.0+2.4.0 libpri-1.4.11.5 I copied my config back into place and most seems to work, but I cannot get my phone that is plugged into the Wildcard

Hello, I had noticed strange crackling sound on my phone calls going through my zaptel device (TDM400P), so i decided to check on possible timer issue, and found lots of issues on forums concerning the sensibility of zaptel with IRQs, and tried about everything: moving PCI slots, noapic and acpi=off boot options, play with different kernel options: iosched/preemption/timer/..., play with