similar to: surpress dial tone on TDM400p

Hi, I recently installed a gentoo on a PC, purchased a TDM400 and installed asterisk. Kernel is 2.6.11-rc3 First, there seems to be a problem with my system: I must put the pci=routeirq to make the card detected. The modules are loaded ok (dmesg output) Zapata Telephony Interface Registered on major 196 ACPI: PCI interrupt 0000:00:08.0[A] -> GSI 16 (level, low) -> IRQ 16 Freshmaker

This is, hopefully, just a case of brain fade. With zapata.conf and zaptel.conf in place, asterisk loaded, no dial plan and all LEDS on the card lit, I get no dial tone, plugging an analog phone into ports 1 or 2, only a buzz and click. zaptel.conf - defaultzone=us loadzone=us fxoks=1,2 fxsks=3,4 zapata.conf [channels] signalling=fxo_ks language=us context=phones-1 group=0

i've tried to set dial = 0/1500 in the indications.conf but still getting a dial tone. is this a bug with *? or did i do something wrong. thank you, patrick

I am having problems getting a Nicherons TDM400P wildcard clone to dial out. Everything appears to be configured correctly, but although I see call progress, it never seems to actually pick up the phone. (The following is a test of 911 emergency, where I substitute 811 [repair service] as the actual number dialed.) *CLI> -- Executing [911 at from-internal:1]

Run /usr/src/zaptel/ztmonitor 32 -v And adjust your gains in /etc/asterisk/zapata.conf accordingly. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of bam Sent: Friday, May 07, 2004 3:35 AM To: asterisk-users@lists.digium.com

I am surprised you needed to turn the rxgain down so much, usually it is just the opposite. I experienced the same problem you did when my txgain was too low. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of bam Sent: Friday, May

Hi, I have a running asterisk 1.2.1 (bristuffed) with a TDM400 Board. Connected to 1 port is am ordanary Fax Machine. Everything 'seems' to work, however receiving faxes is very unreliable. Sometimes I receive a normal page, without problems. Sometimes half of a page and the rest is scrambled, but most often, I receive nothing and the other site reports a Fax error... The Fax machine

Hi, I have a TDM400P (31B) in a PIV 2.8, 512Mb ram, CentOS 4.3, zaptel 1.2.5 and Asterisk 1.2.7.1 and a couple of standard analog phones with a flash button. A hook flash works fine for setting up a 3way call. But pressing the flash button doesn't do anything. The zapata config is below. Anyone have an idea what I'm doing wrong? [channels] context=local usercallerid=yes hidecallerid=no

Hi, I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI

Hi- I'm totally stumped configuring my TDM400P with one FXS and one FXO module. Before I got the FXO module, I used to have an X101P, and everything was working very well. Now * doesn't seem to recognize the FXO channel. I've searched the wiki and the list archives. Stock Debian 3.0 stable installation. Any advice? Thanks. -- David Here's my configuration: modprobe zaptel

Hello, I just setup the Asterisk to integrate with Panasonic legacy PBX. Config as followings, PSTN <-- PanasonicPBX--TDM400P(FXO)--AsteriskPC --> Internet * is for AA / Voicemail and VOIP in/out Currently the AA / Voicemail function for incoming PSTN calls are working well. My problem is for the incoming VOIP call. It can ring my internal extensions and talk without problem. But

Hi all, I'm trying to install a TDM400P card, and I need some help. Please, see below... after dmesg command: [root@voip root]# dmesg via82cxxx: board #1 at 0xD800, IRQ 5 Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 3 for device 00:09.0 PCI: Sharing IRQ 3 with 00:10.1 Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1:

I installed the TDM400P and installed it on a system running on Fedora Core1, please refer to the steps below. I connected an analogue phone to the FXS port. When I pick up the speaker I don't hear the dialtone although when I press a number key I hear a DTMF generated. What could be the problem? I appreciate your help 1- First I installed the TDM400P in the PCI slot and connected

I've setup * with TDM400P w/1 FXS, 3 FXO modules. I've one analog phone connected to TDM400P FXS module, 1 PSTN line to one of the FXO module(ZAP) , and 2 analog phones connected to Sipura 2000 (SIP). The calls between SIPs and zap phone (fxs) are OK. But 2 issues cannot be solved: 1. To dial to PSTN via zap phone, the setup in extensions.conf with the following exten =>

Can anyone suggest where I might begin looking for an answer please? I have just installed a TDM400P (2x FXS and 1x FXO modules installed) The first problem is that it does not seem to be able to detect if the remote party has hung up when a call comes through on the FXO. For example, if someone calls in, and then hangs up at any time after it starts ringing, Asterisk carries on as though the

Hi to all, I have been using Wildcard TDM400P with four fxs modules and it was working fine now I have added another Wildcard TDM400P with four fxs modules . So there are total 8 ports for 8 hard phones. I have modified following configurations In /etc/zaptel.conf loadzone=us defaultzone=us fxoks=1-8 in /etc/asterisk/zapata.conf context=headoffice signalling = fxo_ks

Hi, I have configured tdm400p alone and it works fine for me. But now i tried to add 2 x100p in the same machine with the following configs, but asterisk refuses to load, can someone guide me what i have done wrong. Following r the configs and output. TIA, Azher >> ./asterisk -vvvvvc [chan_zap.so] => (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found --

Dear All The setup is te110p with an 8 channels PRI to make and receive all calls. SIP phones throughout the company. TDM400p with 4 FXS modules to send/receive faxes and make credit card transactions. I have an analogue phone on the tdm400p for testing. I can receive calls to the exten. There is a dialing tone. However, when I try to make a call I get a busy signal. Asterisk stated busy then

Hi all, I'm trying to install a TDM400P card, and I need some help. Please, see below... after dmesg command: [root@voip root]# dmesg via82cxxx: board #1 at 0xD800, IRQ 5 Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 3 for device 00:09.0 PCI: Sharing IRQ 3 with 00:10.1 Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1:

New system install. At what point, in bootup, should I be able to get a dial tone on the phone ports on a tdm400p? There are two fxo and two fxs ports. I know which to plug into <g>. At boot up, as soon as wctdm is loaded, all the ports "go green, yet I do not get a dial tone on the phone ports. I thought as long as zapata.conf is correct, the board should be