similar to: Missing CLI

Hi,   I'm using lars package to run some regression analysis and my doubt now is how can I predict my model to another dataset? Let me explain a little better: I have a dataset from which I withhold some data. With the data that wasn't withheld, I create the model. Now, what I'm not being able to do is apply the model back to the data that I withheld. Any suggestions?   Here it goes

Heres a weird one... Call comes in on mISDN channel. Little bit of dialcode (in a macro) looks up the number in the astdb and puts an name to it. No real magic there, and it works well. Same macro also has parameter passed in to put a prefix on the name - this is set in the DDI handling and is dependent on the number called and allows phone users to see which number was called (company

Colleagues: I have encountered the following situation: SERIES OF COMMANDS source("File1") MORE COMMANDS source("File2") Optimally, I would like File1 and File2 to be merged into a single file (FileMerged). However, if I wrote the following: SERIES OF COMMANDS source("FileMerged") MORE COMMANDS I encounter an error: the File2 portion of FileMerged

Hi, I have 6 Red FXO with TDM2400p in my PC. I have install asterisk and dahdi driver. Scenario is jitsi-----> asterisk server-----> analog PBX ----> landline phone I configured this scenario as follow in chan_dahdi.conf file ; General options [channels] usecallerid=yes hidecallerid=no callwaiting=yes threewaycalling=yes transfer=yes

Preparing to use * for a 'real' installation shortly. Meanwhile, I've got a single port clone thing, 00:06.0 Communication controller: Motorola Wildcard X100P working to answer my landline and send calls to my laptop or voicemail. Sweet! Trying to "call out" from linphone, I set up this: exten => _X.,1,Dial(DAHDI/1,${EXTEN}) Both SIP client and this extension are in

On a BT line, how do I determine whether the number on an incoming call has been deliberately withheld (by dialling 141) or is merely unavailable (e.g. because it originated from overseas or passed through some ancient switching equipment) ? In the first case, I want the caller to be played a message to the effect that we are not at home to anonymous cowards but if their business is

I have Asterisk 1.4.24 en a Digium TDM410 with EC and with 3 FXO modules. When I plug one PSTN-line into a FXO-port I am able to receive calls on this line and I can also make calls from an internal SIP-phone to the external PSTN-network. Still I am bothered about something that appears on the CLI when I do a reload chan_dahdi.so : asterisk*CLI> reload chan_dahdi.so -- Reloading module

Hi, We are running 1.4.22 and have been experiencing problems with certain IVRs and DTMF Tone duration. We would like to be able to increase DTMF Tone duration by 50 to 100ms over what the user is pressing on his phone. We have a PRI test circuit and an analyer in between to measure tone duration. We have tried setting chan_dahdi.conf parameter 'toneduration', but that does not do

This is a real 'newbie' type question, but I can't get my brain to work today. Is it possible to re-direct an incoming SIP call based on it's CLI? Ideally I would like to check incoming calls against a short whitelist (of say 20 numbers) and redirect to a different extension if there is a match. I would also like to redirect calls that fail to present any CLI (aka

Hello I have contacted my line provider who is saying that in order to get my 0845 or 0870 number to id as the incoming number on a landline that i may call i need the following. User must provide - NPI set to E.163/E.164 User must provide - TON = "national or international I have had a good search around and can't seem to find a good answer to this. Does anyone have any idea where i

Hi all, I tried to set the calleridname of an incoming call to get different incoming labels displayed for different incoming numbers. This does work for hidden number-calls so I can set the displayed CIDName on my cisco7960 from "CID withheld" to "abc CID withheld" If the incoming CID isn't hidden it works to use SetCallerID but not to change only the CIDName with

Hi, now I've got caller id working on my BT line in the UK, I'd like to play a different message to those pesky sort who with hold their outgoing number. How can I do this in my extensions.conf for my [incoming-analog] context? I realise some people may call who are unable to change the way that their system withholds the outbound number, so I'll give them chance to leave a voice

Hello Team I have installed the new DL580 and used the new TE420B to add capacity on our ivr. Before I put new E1?s I decided to first move the old e1 from the old system to this new one but it has errors which not only affect the audio quality, but also cause the asterisk to refuse any call after sometime even though the channels seems up and active {seems d-channel fails}.. When processing

Hi, Starting in Asterisk 1.8.0, Asterisk supports connected line updates. This is fantastic for SIP. How can I prevent them from being sent to a PRI channel? I'm having problems when a call is answered by an internal SIP extension, then transferred (blind or attended) to another internal SIP extension. One of my PRI providers can't handle the ROSE_ETSI_EctInform APDU and drops the

Hi all, I downloaded a copy of bristuff 0.4.0-RC11 on my Ubuntu 10.04.2 LTS server machine with 2.6.32-24-generic-pae kernel. The prereq.sh script executes without complaints (BTW on my system, libncurses-dev evaluates to libncurses5-dev and libz-dev evaluates to zlib1g-dev). With the asterisk 1.4.41 package that is installed, a make menuselect operation indicates that all dependencies are met

Hello, I seem to be having an issue with the configuration of my PRI on a new asterisk server I've created to replace an old install that I have. The card is Digium Wildcard TE133. I continually get messages like "Primary D-Channel on span 1 down", rather irregularly: [2013-07-29 17:31:39] VERBOSE[3621] sig_pri.c: == Primary D-Channel on span 1 up [2013-07-29 17:31:39]

Hi list! We currently have a PRI gateway composed by a box with two Digium quad-span PRI cards (a TE420 and a ). One of the cards is filled with TELCO1, while the other has first two slots filled with TELCO2, and 3rd slot with TELCO3. I am currently having (timer ?) issues on TELCO3 (span 7) D-Chan (202 as determined by dahdi_genconf ) is constantly failing causing on-going calls to terminate.

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> Hi everione,<br> <br> I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri 1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the timer fail with a telco

We fairly recently switched service providers for our 4 PRI circuits. Since that time, we started to notice some failed inbound calls. These calls terminate with an ISDN cause code 47 'resource unavailble'. Most of the time I see this error on the first or second channel on the second span in a trunk group (This is the providers trunk group for hunting, not an Asterisk trunk group). All

Hi, I'm in the process of switching over to Asterisk from Alchemy kit and have hit a stumbling block. We're in the UK and use ISDN. At the moment we don't accept calls from withheld numbers (we just play them a message), but do accept calls from unavailable numbers. There doesn't seem to be any way for me to differentiate between the two number types in Asterisk (chan_CAPI) -