similar to: Problems with Module

Hi there, I've got two Asterisk hosted PBX servers with Digium TE210P cards attached by a E1 cable to Port 1 on each. On startup, both cards flash red, alternating between ports 1 and 2. When server #1 loads the Zaptel module and drivers, Port 1 status LED goes green. When server #2 loads the same module and drivers, Port 1 status LED goes completely blank. Unloading the wct2xxp

Hello everyone. I have server with 2E1 PCI card, asterisk 1.4.35, dahdi 2.4.0, libpri 1.4.12-beta2. One PRI trunk looks to PSTN and take a clocksource from telco. Another trunk looks to PBX with DECT system. Some outgoing calls from asterisk to PSTN drops. The last message that exists before hanging up process is: DEBUG[28467] channel.c: Got a FRAME_CONTROL (8) frame on channel DAHDI/... This

Hi, I'm trying to get a TE212 working on a Dell PowerEdge 1850 running Debian etch using the latest release of libpri (1.4.0), zaptel (1.4.2.1) and asterisk (1.4.4). The initilization of the Octasic echo canceller seems to fail when the wct4xxp module is loaded. [...] VPM450: echo cancellation for 64 channels Failed to open chip, code 00103017! VPM450: Failed to initialize [...] By looking

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

Hello, I'm running Asterisk@home 2.5 asterisk 1.2.4 zapatel 1.2.2 libpri 1.2.2 on a Dell Poweredge 2850 (1 CPU) with a TE210P I have 2 t1 circuits using NFAS with dchan on 24 and no backup dchan. I am able to receive inbound calls on all channels and can only make outbound calls on channels 25-48. Attempting to make an outbound call on channels 1-23 results in congestion.

Hi, I'm running Asterisk 1.4.2 on a Debian GNU/Linux and a Digium TE210P card. I have one E1 with MFCR2 Signaling. I compiled asterisk + libunicall, and I can make calls over E1, but can't receive. Here the CLI when I make a call: -- Executing [006191642208@ps5:1] Dial("SIP/23-081cbc40", "Unicall/g1/91642208|50") in new stack -- Called g1/91642208 [May 14

/bin/sh: -c: line 0: syntax error near unexpected token `;' /bin/sh: -c: line 0: `if [ -n "" ]; then if [ -f ]; then mv -f .bak ; fi; cat .bak | grep -v "alias char-major-250" | grep -v "post-install torisa /sbin/ztcfg" | grep -v "post-install wcfxsusb /sbin/ztcfg" | grep -v "alias wctdm" | grep -v "post-install wctdm

I am connecting 1.4.22 and dahdi 2.1.0.3+2.1.0.2 to a siemens hipath 300 and siemens hipath 4000. (2 channels to each switch) with a TE210p card setup as T1 with em_w. When the call is initiated to either switch the phone rings, when its answered then nothing... I hear no audio etc... After the timeout period the call is hung up. The phone switch 300 needs the T1 reset as the channel is not

Hi all, I'm going to configure a middle asterisk installation. I'll use a TE210P to connect a T1 channel bank and a PRI E1 line. I'm thinking on using a SuperMicro P8SCT Mother Board that has a 1x 64-bit 133MHz PCI-X 3.3V. In TE210P documentation I've read: The TE210P is a 32-bit 33MHz card keyed for 3.3 volt operation. Can SuperMicro slot (that is a 133Mhz slot) be used

We are running Asterisk version 1.4.23-1, libpri-1.4.9 and zaptel-1.4.12.1 and two Digium TE220Ps. Debugs are set to 10. We have a T1 PRI connected to the telco. Over the last 4-5 days, we have getting Yellow/Red alarms coming from the T1 PRI. The other two ports in use are connected to internal test switches (Avaya Legend/Avaya Definity), and are not showing any errors.

Evening, My latest asterisk box is having a difficult problem. It is configured with one TE210P and TDM400P with four FXO modules. I'm running FC6. The TE210P only has a single PRI. When the system boots, it is completely random what order the zaptel modules will get loaded in. Sometimes zttool shows the FXO as the last span, sometimes as the first. When it does load as the first, which

Hello all. I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box. NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco Incomming calls from the telco to the asterisk box to the NEC work fine with indials and everything. Works sweet. Outbound from the NEC to the Asterisk box fail. Giving an long dial tone that then times out. Ie, pick up NEC handset, dial

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I am stuck in a situation where I have two te110p's and really need a te210P. Anyone interested in a trade? They are in working condition guaranteed. If this is the wrong forum for this, I am sorry... I will take it off list. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Thunderbird -

Hi list! I'm trying to setup a new * server with a TE210P in a Dell 2850 box. I'm using zaptel version 1.2.5, the linux flavour I'm using is CentOS 4. In a previous thread I read about the results I should expect from zttest. On my home box (using the crappy Asus A7V600) I got really bad results from zttest (just over 97.5) but I know that this motherboard just sucks. To my

Hi all, Our asterisk PBX, randomly restarts all the channels of the E1 connection. It sends this message "There is no D-Channel, using channel 16 anyway".Then the asteisk recive (or it thinks it recives) yellow alarms at all the B-channels, after that it restart all the channels. When restarting the B-channels it cut all the conversations that is handling at that moment. Does anyone

hi i'm trying to configure a TE205P on asterisk@home i've edited /etc/sysconfig/zaptel adding this line: MODULES="$MODULES wct2xxp" now, when the system is loading, i can see that the wct2xxp module is loaded correctly but if i try the command: /usr/local/sbin/genzaptelconf i get: STOPPING ASTERISK STOPPING FOP SERVER Generating '/etc/zaptel.conf' Generating

Hello, I'm just starting out with asterisk and I'm playing around with the system. Currently I have a Digium TE210P connected to a PRI on the Asterisk server. I have a SIP soft phone on my laptop for testing that is working fine. When I try to place a call from my soft phone I get this from Asterisk: May 1 09:11:41 NOTICE[20098]: app_dial.c:1029 dial_exec_full: Unable to create

Dear all I have now install TE210P 2 port E1 card on asterisk 1.4.10 on centOS 5 but thing is that i have connect 1 E1 port with avaya E1 back 2 back and second E1 card on Direct Telcom for outgoing for outside now i got this error when i call on avaya PRI asterisk think PRI_CPE and remote end also CPE i have configure /etc/zaptel.conf span=1,1,0,ccs,hdb3

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Anyone have a 1650 running successfully in production mode with 2-4 PRI's? I want to make sure I don't have a motherboard compatibility problem before I buy one of these. We are going to be using a Digium TE210P to start off with and probably moving to the TE411P down the road aways? thanks, Sean -----BEGIN PGP SIGNATURE----- Version:

Has anyone every attempted to set up a mini PC to achieve much the same functionality as the fonebridge box? The sort of thing I'm imagining is a micro itx board & case in a completely solid state configuration (flash disk, maybe a psu fan but only if really required), with a TE210P (or equiv) card(s). The sole purpose of this would be to be a bridge between E1 lines and TDMoE. The