similar to: transferring call to another line

Hi, I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxs (for the fxo cards), which worked fine. ztcfg is showing two channels configured, but when I start asterisk and do show channels, i see no active channels. zapata.conf has: signalling = fxs_ks

Hi, I've been trying to get my zaptel x100p cards working for the past week now. this is what I've done: installed asterisk: make clean make linux 26 (for fedora core 2) make install installed zaptel: make clean make make install did a modprobe zaptel, and wcfxo got this in /var/log/messages: PCI: found IRQ 11 for device 0000:00:0f.0 wcfxo: daa mode is 'FCC' found a

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

I am trying to configure Asterisk for BSNL, india network. I have successfully configured it for outgoing calls. When any outside number make any call to trunk then it receives the call properly but when the call is disconnected by inside extension then outside phone does not get a busy tone. Asterisk incoming call log: -- Executing [s at incoming:2] Dial("Zap/4-1", "Zap/1")

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

Hi, I recently purchased two X100Ps and a TMD40B. I have set everything up according to the instructions however I can't get asterisk to pick up an incoming call (It just keeps ringing). Also, when I pick up an internal extension I get dead air. I searched the mailing list for 3+ hours -- What am I missing? Thanks in advance for your help. Here are the installion steps that I used:

Well, I thought I had my problem solved, but it is acting up again. Hopefully this time I can provide enough information. What I have is an * box setup with one X100P and TDM400 with one FXO and one FXS. For my regular setup with interfacing with my PSTN and my entire house with analog phones, the box is working great. I am trying to interface a Mediatrix 1202 device to my * box via the

Hello, I have a TDM400 with 2 FXS modules and 2 FXO modules, as follows: Zap/1 - internal phone Zap/2 - internal modem Zap/3 - exchange Zap/4 - exchange Recently I upgraded zaptel (and kernel modules) from 1.0.9-1.1 to 1.0.9.1 and upgraded asterisk to 1.0.9.dfsg.1-3, which are Debian versions for Debian/sarge of the packages I believe the Debian maintainers created (not in Debian) The above

We're having issues with the FXO/FXS ports on our Digium TDM cards sporadically. I'm wondering if anyone else has had these problems, or if anyone can provide guidance diagnosing or fixing the issue? The symptoms are that the FXO and FXS ports "stop working", usually after 2-4 weeks of server uptime. When this happens, sending a (SIP) call to an analog phone on an FXS port

Hi there, I have a TDM400P (1 x fxs, 1 x fxo) which I'm attempting to run on linux 2.6 (gentoo), without much success at the moment. I have previously had it working on a 2.4 installation, but when I moved to a new box and installed a 2.6-based system, it failed to work. In both cases I'm using whatever (libpri, zaptel, asterisk) is checked out by default (I assume that means HEAD)

I've tried to ask this before but didn't want to waste a lot of bandwidth. However, I have to give the params in order to make the exact question understood. We have 2 ZAP FXO connected to two phone lines(2xX100P) 3 phones connected to a TDM400P with FXS modules Usually a caller will be on ZAP/2 and the first outbound line in the g1 group is on ZAP/1. There are two (main) call waiting

Hi, in response to a previous posting regarding getting the x100p to work, I was told to run "zap show channels," but when i do i get "no such command 'zap'" There was a previous posting on this, but the guy never posted the solution. thanks, Imran

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

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,

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 need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal ---------------- No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone,

I'm using two Rhino channel banks (first 12FXO/12FXS, second 24FXS). These connect to a Digium TE210P card. I'm running kernel 2.6.10 and I've tried Asterisk (w/zaptel) 1.0.9, 1.2 beta, and CVS from today. The results are the same for all versions: Right after I reboot, and modprobe wct4xxp, my analog phone connected to port 13 of the first channel bank (first FXS port) gets a dial

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, 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

Dear List; Thanks alot for the help. But how can I let the second dial tone (after pressing the extension to select that FXO port) to be difference than normal dial tone? Regards Bilal Ghayad -------------------------- Correction, on FXO port not FXS, second, read his email first: "Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP