similar to: Someone please explain FXO/FXS module/channel relationships in Zaptel configuration to me

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 am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS modules. I'm trying to set-up things to route analog fax calls from a FXO port to an analog fax machine on a FXS port on the same card. Outgoing faxes work just fine. But incoming faces are routed to the right DAHDI extension, but the call dropped right as the fax machine rings for the first time. The fax machine

I'm having some trouble dialing phone numbers in Argentina with Digium Zaptel cards. Does anyone have some sample configuration that works with Digium TDM04B cards in Argentina? I'm mainly referring to the /etc/zaptel.conf and /etc/asterisk/zapata.conf. I have two Zaptel cards: the first one has 1 FXS port and 3 FXO ports; the second one has 4 FXO ports. My current configuration is

I'm a bit confused on how to setup Zaptel.conf and Zapata.conf when there is a TDM400P and a TE410P installed after upgrade. The TDM400P has 2 FXS in position 1 & 2 and 1 FXO in the fourth position. I see boot, WCT4xxP loading first and WCFXS loading second. According to my understanding, given above, the TE410P should be configured first, then the TDM400P. However, I'm not sure

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]

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

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:

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

I have a TMD400 card installed in a PC with one fxs (installed in slot 2) and two fxos (installed in slots 3 & 4).? fxos work fine but I am unable to get a dial tone for any devices connected to the fxs.? I have correctly connected the power supply to the card and I have even tried moving the card from slot 1 to 2 on the board. Below is from the console when I try to route a call from FXO on

Has anyone got a sensible zaptel.conf and zapata.conf for 2 TDM400P's working with UK set-up. They're set-up with 7 analogue phones and 1 PSTN port. Currently zaptel.conf has fxoks=1-7 fxsks=8 loadzone=uk defaultzone=uk It's really zapata.conf that would be useful. Currently using the zaptel/asterisk that comes with Ubuntu (latest) which needed a bit of tweaking (1.2.16), but could

Hi, We have a PRI Trunk (physical E1) and we are getting some rather weird and very isolocated problems. On outbound calls to specific numbers, it would seem to me that DTMF from the remote side is affecting the local asterisk system. Basically what happens: - We make a OUTBOUND call via the PSTN (PRI Trunk) to a remote System - Remote Answers, and converse - Remote sends DTMF on their site to

I got this nice TDM22B with two green modules left and two modules right /var/log/messages shows: Nov 26 23:47:48 dns kernel: Zapata Telephony Interface Registered on major 196 Nov 27 00:37:53 dns kernel: Freshmaker version: 71 Nov 27 00:37:56 dns kernel: Freshmaker passed register test Nov 27 00:37:56 dns kernel: Module 0: Installed -- AUTO FXS/DPO Nov 27 00:37:56 dns kernel: Module 1: Installed

I got this nice TDM22B with two green modules left and two modules right /var/log/messages shows: Nov 26 23:47:48 dns kernel: Zapata Telephony Interface Registered on major 196 Nov 27 00:37:53 dns kernel: Freshmaker version: 71 Nov 27 00:37:56 dns kernel: Freshmaker passed register test Nov 27 00:37:56 dns kernel: Module 0: Installed -- AUTO FXS/DPO Nov 27 00:37:56 dns kernel: Module 1:

Hi, I'm installing an Asterisk box with a TDM2400P in Brazil. I can make analog phones work while lines are not working. Since I do not know anything about brazilian lines, is there anybody who can tell me what is wrong/missing in my conf files (below)? TIA Giorgio _zaptel.conf:_ fxoks=9-16 fxsks=17-24 defaultzone=br loadzone=br* * _zapata.conf:_ context = inbound_zap echocancel = 128

Hi, I know it sounds weird, and this is part of the reason I have not reported that sooner. As I upgraded from 1.6.2.x to 1.8.x several months ago I am experiencing this problem. If a call is initiated from a DAHDI extension after no DAHDI extensions were used for some time, arbitrary DTMF digits are skipped and the call fails. If the call is redialed it goes through. Normally just one (1)

I have a Digium TDM400p (1 FXO, 1 FXS) with the FXS module connected to a standard analog handset with caller id display (US caller ID). Although it appears that caller id information is coming into asterisk (it shows up in voicemail), I can not get it to display on the analog handset. Is there anything special I need to do to send the caller id info out the FXS port? I've tried a few

Looking for some help from the UK please. I backed up all my Asterisk configuration before re-installing the server from 32 -> 64 bit. Unfortunately I did not transfer the backup to another machine!!!!! I now have a TDM400P that is not picking up the line. Can you see what I have done wrong when I have rebuilt the config please: dahdi_scan ---------- [1] active=yes alarms=OK

Hello list , I?m putting to work a new asterisk box. I?m running * 1.0.3 with (one ) Wildcard TDM ( 2FXS* 2FXO) and (one) E-100P. Both boards are working well. ztcfg don?t show me any error. zttool list both cards as "OK" But when i run asterisk on verbose mode i get those errors : [chan_zap.so] => (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found --

Hi, Would someone assist me with the configuration of my TMD10B and X100P cards? I would ask Digium but my support was good for only one install...( I had to rebuild my * box). I wrote down the configuration but I must have made a mistake somewhere. I have followed the instructions on the Digium site but I must be missing something. Here's the config (snippets) I've been attempting to

I have a bit of a weird problem that I'm having great trouble debugging. I have a TDM400P PCI card with two FXO and two FXS modules. Both FXO modules are connected to BT lines here in the UK. Both BT lines have V23 Caller-ID, which works fine with Asterisk. Both asterisk and zaptel are fresh from CVS. Both FXO modules (channels 3 and 4) are in "group 1" for outgoing calls. My