similar to: TDM400P REV I issues - ProSLIC vs TDM400P

Hi, I have received a new card TDM400P revision E, from digium. When I tried to modprobe wcfxs it gave me the following errors: Freshmaker version: 63 Freshmaker passed register test ProSLIC on module 0 insane (1) 255 should be 2 Module 0: Not installed ProSLIC on module 1 insane (1) 255 should be 2 Module 1: Not installed ProSLIC on module 2 insane (1) 255 should be 2 Module 2: Not installed

Hello, I have few questions about Asterisk. I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago. 1.I couldn't find Asterisk version using "asterisk -V" command. How can I to find version information? 2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on it. I tried Asterisk CallerID feature, but unable to get it. I tried callerid signaling V23,

hi guys, i got a TDM400P FXS card an everything is fine except for this when i do modprobe wcfxs , the linux shows 2 TigerJet Network Inc Model 300 128k. i don't know why it is showing 2 of them. or is that what it is? ERROR: Freshmaker version: 63 Freshmaker passed register test ProSLIC on module 0 insane (1) 0 should be 2 Module 0: Not installed ProSLIC on module 1 insane (1) 0 cshould be

Hello, I have few questions about Asterisk. I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago. 1.I couldn't find Asterisk version using "asterisk -V" command. How can I to find version information? 2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on it. I tried Asterisk CallerID feature, but unable to get it. I tried callerid signaling V23,

I apparently have a dead FXS module. Is there any kind of test I can run on it (on a live system) to determine if its good or bad? The telephone has no dialtone, gets no calls. It has been working for several months, and just quit yesterday. Thanks! Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO

Hello. I have this problem. In my asterisk box, I was running debian woody with asterisk package from backports.org. Last friday I upgraded from debian to sarge and change from kernel 2.4.18-1-686 to kernel 2.6.8-1-686, rebuild zaptel kernel module and also upgrade to asterisk 1.0.2. But now asterisk won't start. Here is more info #asterisk -vvvvvvvv (last lines) [chan_zap.so] =>

Greetings all, I'm new to asterisk... Finally got two IP phones and two TDM400P's (one with four FXO modules and one with four FXS modules)... Began experimenting a couple days ago and have ran into a very aggrevating problem. My system consists of the following hardware: Abit BL7-RAID motherboard Pentium 4 1.6GHz processor 512MB SDRAM 2 TDM400P cards 2 3Com 3c905

Hi, I'm trying to get up and running a TDM400 with a standard italian pots line but i'm having problems at getting asterisk to detect when the line get answered on outgoing calls. I'm using asterisk 1.6 beta 9 with zaptel 1.4.11. I tried with and without answeronpolarityswitch=yes but it didn't change anything at all. With callprogress=yes answer get never detected. With

Hi all, Does somebody know why no load modules to FXS? I used zaptel-1.0.7 version driver. [root@server1 zaptel-1.0.7]# modprobe wctdm ZT_CHANCONFIG failed on channel 3: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? /lib/modules/2.4.21-27.EL/misc/wcfxs.o: post-install wcfxs failed

Greetings, Somewhere on my asterisk system, a calls come in in a certain context, for example, from-sip or from-pstn. Then the calls gets routed through the dialplan, and a macro gets called, and another one and then the call needs to be redirected to another number in the same initial context. And you can use Dial(Local/number/initialcontext) for that. Oops, this initial context is lost

Hi, I am interesting how can I use the capabilities of ProSLIC to measure the following PSTN parameters: - Voltage (V) & Polarity (+-); - Current (A); - Frequency (Hz). Are there any ready for use tools? If there aren't ready for use tools how can I do the above measuring? Best Regards, Miroslav Nachev

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:

Hi, We have two of these 4 port FXO cards. However, we are having some problems with incoming/outgoing calls. The latest version of Asterisk/zaptel from CVS is being used. Voicemail, internal SIP <-> SIP calls between Pingtel xpressa hard phones work terrific, echotest is fine, and so on. The zaptel and wcfxs modules load fine, and show all 8 FXO interfaces in dmesg:

Hi, For the project I've used the Eicon DIVA card. It has 8 BRI ports, and for about 25% of the time there are 7 or 8 in use. So we want to replace it with an E1 card. Only issue is, replace it with what? The idea we have been playing with was to get a Digium E1 card (we already have bought lot of Quad E1 cards :-) and then just put it back to back against Asterisk server. And instead of

> 3. Supposed I have 2 fxo cards (right now I have one already) and 3 > fxs, and one of the fxo will have two phone (running pararell), is > there any way for * to: > a. It always dial the first fxo, if the fxo is busy or is being used > (have other people conversation), will * be able to switch it to > other fxo? Here's the approximiate the conditions of the phone.

I have a TDM400P with one FXO module and two FXS modules in it. I also have a Wildcard X101P. After trying hard to get things working on various Intel computers, but having echo problems that made it not really usable, I decided to try it on some older PowerPC (Macintosh) hardware running Yellow Dog Linux. Things started off smoothly. Both zaptel and asterisk seemed to compile okay, and both

This afternoon I finally figured out more with regarding to a strange clock-slip problem we have on our asterisk box. We have two TE410s, in E1 mode: TE410P version c01a009b They have their own interrupts: 66: 781648298 783747388 IO-APIC-level t4xxp 233: 253890977 1311504670 IO-APIC-level t4xxp They have their full 31 channels: span=3,0,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78

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

Hi all, Yesterday, in an attempt to take back my phone room, I pulled everything apart as far back as the demarc and rebuilt it. In the process of putting things back together I accidentally connected my incoming lines to my FXS ports and my phones to my FXO ports. I quickly realized the mistake I made and corrected things but not before one of my FXS modules was smoked by incoming ring voltage.

I have seen that file ftp://ftp.openbsd.org/pub/OpenBSD/OpenSSH/portable/openssh-3.4p1.tar.gz is continuosly changing. This seems strange to me as I expected it should be a 'frozen' file; The signature and diff file are still dated Jun 26. I am wondering whether this is intentional. Best regards, Wojtek