similar to: t100p configuration troubles

In my * server I have a T100P, connected with a T1 crossover cable to an Adtran TA750 with 2 FXO cards. After I do 'modprobe wct1xxp;ztcfg' the alarm on the card starts blinking red. Running zttool confirmed that yes, it is a red alarm. I've read a few previous posts on this type of setup and I'm not too much closer. What should I be checking next? A different cable? The

(to mods: sorry for any extra work I've put you through. Posting from my work account, as proper) Hello all, I'm back. I've got that lucky feeling that this will all work out very soon, and I hope that that is true. The issue: In my * server I have a t100P, connected with a T1 crossover cable to an Adtran TA750 with 2 FXO cards. After I do 'modprobe wct1xxp;ztcfg' the alarm

Hey all, I've setup * to serve the needs of our small helpdesk and I'm looking to expand. We're planning on doing support for different companies, each one identified by a different 1-800 number that terminates at our PBX. What I would like to know is: is there a variable I can read to determine what number any given caller dialed? I'd like to be able to separate calls based on

Hi, Both under the current R-devel (r66456) and a version from about 3 months ago, I experience the following behavior: > parallel::detectCores(TRUE) Error in system(cmd, TRUE) : error in running command > traceback() 3: system(cmd, TRUE) 2: gsub("^ +", "", system(cmd, TRUE)[1]) 1: parallel::detectCores(TRUE) > This is on Ubuntu 14.04. Does anybody else see this? [I

Hello, We've released another update to our Asterisk GUI Client suite: http://astguiclient.sf.net/ Screen shots: http://astguiclient.sourceforge.net/screenshots.html The client suite runs on both Windows and UNIX and includes a dialer (the suite is not an asterisk configuration tool) In addition to the usual bug fixes, this is mostly an update for the VICIDIAL dialer application.

Hi! I want to install a X100P. I think I did everything according to the manuals I found in the net. I loaded the modules and I edited the config files according to <http://www.digium.com/downloads/hw_article>. I start ztcfg that tells me everything is alright. But when I start asterisk it tells me the error as posted in the subject. Why? I'm using kernel 2.6 could that be a

Hi, I'm trying to make multiple poisson regressions with the MLE2 command. I have used the following expression, but I receive an error message: poisfit <- mle2(y ~ dpois(exp(b0 + b1*x1 + b2*x2)), start=list(b0=1, b1=1, b2=1), data=data1) Error in optim(par = c(1, 1, 1), fn = function (p) : non-finite initial value 'vmmin' I have changed initial values using coefficient values

Is anyone successfully using call queues and call groups? If so do you have an example configuration? The wicki and mailing list information I found is pretty old. Thanks! Paul pmahler@signate.com

Greetings, I am attempting to configure an AS5300 to provide a SIP based gateway to the PSTN from Asterisk. I have been unable to identify through the docs how specifically this should be configured in Asterisk and have not been able to get things working through trial and error. I am sure I am missing something fairly obvious here but any guidance (or example cfgs) would be much appreciated.

I have a small HUGE problem with *. I have installed * but I have 2 problems. 1 - Making call to FWD. 2 - Receiving call from FWD More info of the problem at the end. Here is the sip.conf file. ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls register =>

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

I'm trying to get two linux machines with kphone-3.11 two communicate with each other over asterisk. I have them configured correctly on asterisk to use sip channels, but when I call from one phone to the other I don't any voice communication between the phones. According to the phones I'm connected, but according to asterisk, I get the following message: -- Executing

I have the following asterisk configuration (sip.conf) : [General] externip=82.79.81.3 localnet=192.168.10.0 localmask=255.255.255.0 [Phone1] type=friend host=dynamic nat=yes qualify=yes context=sip callerid="Phone1" <1> disallow=all allow=gsm [Phone2] type=friend host=dynamic qualify=yes context=sip callerid="Phone2" <2> disallow=all allow=gsm [Phone3]

Don & others, Thank you for your answer. The fog maybe lifting ;). The zaptel.conf file has the following in its comments: # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of "1". For a secondary, use "2", and so on. # To not use this as a

Dear R-list, I know is going to be weird, but I need to do something about and I know the R-list is big group of good people. I?m moving to Philadelphia in August 2009 (my fianc? is there) and I need to find a job. I know to do so many things, but my skill is in science and research. I would like to know if you know a possible position near Philadelphia, New Jersey or New York. I was working in

Hi, After reading this valuable forum and the voip-info wiki and follow all the steps , but my Cisco 12SP+ remains unregistered. These are my config files: skinny.conf [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 172.20.1.1 ; Address to bind to dateFormat = D-M-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 languaje=es allow = all ; disallow

Hi Folks, I have a T100P ordered from Digium in the past. Its working perfectly fine but I now need to more the server to India (out-sourcing) and we will be getting an E1 line there. Does anyone know if T100P was capable (upgradeable) or swappable so that I could use it with the E1 lines instead? Any pointers to driver/firmware update would be highly appreciated. Thanks. R -------------- next

I'm setting up a T1 for both voice and data and was wanting to know the steps involved. I've recompiled zaptel for ZAPATA_NET support and setup my /etc/zaptel.conf: Fxsks=1-10 Nethdlc=11-24 Modprobe wct1xxp works fine and my voice lines work. Now when I use the provided sethdlc, I use the following commands: Sethdlc hdlc0 mode ansi Ifconfig hdlc0 <ipaddress> <network>

I got it figured out now. In case anyone else comes across it - this is what I had to do: Sethdlc hdlc0 mode ansi Ifconfig hdlc0 up ***NO IP INFO SET*** Sethdlc hdlc0 create <dlci number> Ifconfig pvc0 <local ip> pointopoint <remote ip> up Then set your default gateway to the <remote ip> listed in pvc0. That's about it!! Hope it helps! -----Original

I ran into a couple of odd problems with a new Asterisk box I have set up... I have a T100P connected to a channelized T-1 (24ch) to the PSTN. The T-1 is set up properly so far as I know and seems to work perfectly with one exception: When a call comes in, asterisk starts a 'simple switch' for it, goes through my 's' extension progression until it gets to the first Playback