similar to: zapata initial context question

Another dumb question... My asterisk system seems to have problems detecting hangups. I am getting a LOT of voicemails with dialtone or silence. I see over at asteriskguru.com there is an explanation of how to configure for polarity reversal in zapata.conf? Does zapata.conf have any function in systems that aren't using zaptel( I suppose not)? I am using an external gateway (wellgate

Hi *, now for a long time i am trying to set the outgoing callerid, without luck. I am here in Germany, my asterisk has a pri interface connected to a PMX installed by Telekom. All telephone calls are preselected to EcoVoice. I am using asterisk 1.2.7.1, zaptel 1.2.5 and libpri 1.2.2. A week ago we tried with a device able to simulate a telephone system so send out a callerid, and that

I'm in the process of setting up a 'phone tree', and are running into some problems. My goal is for users to dial a phone number, the asterisk system picks it up, plays the greeting, and users can type whatever they want into the system. What actually happens is users dial the phone number, asterisk picks up and additionally goes off-hook on another line, plays the greeting and

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

Hello all, I've got a T100P connected to an Adtran TA750 with a T1 crossover... This connects to a patch panel with phone ports. The Adtran is fully populated with FXS cards. All I get on any phone port is a fast clicking noise... No dialtone. Asterisk 'sees' the card, (the channels show up in /proc/zaptel). Incoming calls are routed to the zap/x channel, but no ring. I'm

Hello, everyone After having spent several time to look for any solution for my problem, I decided to write here. Here is the problem got a Digium X101 FXO card and Digium TDM400P alias freshmaker (1 fxs module on it on first port ) in my asterisk box. The X101 works perfectly. The problem comes from the TDM400, when i plug a tested working phone on it , i get no dialtone; the goal is to

hi.. nope I do know about x100p being the fxo and the usb device being the fxs ....fxo using fxs signalling and fxs using fxo signalling and it is reflected suitably in my zapata.conf and zaptel.conf... The problem is that the S100U usb device is not identified upon zap show channels ... only the 100xp wildcard is recognized....i get a sound when pressing the keypad of the phone connected to

Hi I have been searching the wiki and mailing lists and I cant see where my config is incorrect. I have a Digium tdm11b (1 fxo + 1 fxs) this is the output of cat /etc/zaptel.conf # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCTDM/0

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

Can someone please help me with my config settings for my TDM12B. I have got the FXS port to work fine, but I cannot get asterisk to startup with any settings for the fxo ports. It appears to be something related to the message "Signalling requested is FXS Kewlstart but line is in FXO Kewlstart signalling". However, I have tried what appears to be every possible configuration option. I

You should go here and enter your suggestion so the developers can track it. http://sourceforge.net/tracker/?group_id=123387 Cheers, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Aaron Glenn Sent: Thursday, March 10, 2005 4:38 PM To: Asterisk@Home Cc: Asterisk Users Mailing List - Non-Commercial

Hi pals, Pardon me if this question sound basic please. This is the first installation where I have to use the analogue card and therefore a little lusty. I have googled a lot, but though there is a lot of information about the above file, none indicate where the file lives. I have a installed asterisk and zaptel software on a fresh installation of CentOS 5. This all from source and following

-----BEGIN PGP SIGNED MESSAGE----- Does anyone have any working example GR-303 configurations for zaptel and zapata conf? The information available on the wiki as well as in the sample configurations just doesn't seem to be enough to bridge the gap for me. In Zaptel.conf, Do you set up a GR-303 circuit like a PRI with b and d channels or do you set fxo or fxs, ks signalling? How do you

Folks, I recently swapped a TDM400 FXS card that was working perfectly into a new server (running recent CVS), and it's either misbehaving (unlikely), or I've missed something obvious (much more probable). Everything seems to be working, but I can't get any dialtone from it when I plug a phone into any of the 4 ports!! All of the jacks are lit (green lights) but they all seem dead

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:

I have recently installed Asterisk Now Beta 4 (32-bit) with three Digium cards - 2 Wildcards TE400P (one with 4 FXS modules and one with 4 FXO modules) and a TE205P dual E1/T1 card. The problem is with the TE205P and configuring the /etc/zaptel.conf and /etc/asterisk/zapata.conf files. I go in and add the required lines to configure the TE205P and restart Asterisk. All is fine until I reboot the

Installing DevkitLite hardware (Very similar to John Lange's post on Tue Oct 08 2002) I cannot get anything to work on the phone connected to the s100u. I dont know what to do. Can someone please help me? I used the sample configuration files from digium documentaion that was supposed to be "sane" defaults for the kit. Very similar to John Lange's post on Tue Oct 08 2002 Here

Hi, I am trying to configure * 1.0.7 box with a Digium Wildcard TE110P T1/E1 and libunicall latest code. All libs compiled successfully and the E1 have a green light! I am able to receive a call (or at least) testcall shows some information when an incoming call is received so the drivers and basic configuration is working. My * box has 2 cards, one TDM400B (2 fxs and 2 fxo) and one TE110P

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

Hi All I am working on asterisk + digium developer card , it has on FXO and one FXS I want to work asterisk in the following way 1>FXO connected to PSTN line 2>the calls coming to PSTN line should be received 3>SPI clients should be able to call outside through PSTN 4>There is no phone connected to the FXS In this case , do i need to bother about configuring FXS , in