similar to: unamble to dialout to mobiles and others "special" numbers

It seems that you have load the "survival" package date.mdy is a function from this one. In this function the "origin" of the time is the first day of 1970 in the base package the origin is the first day of 1960 it's very curious... Benjamin Esterni France From: "Vegard Andersen" <vegard.andersen@ism.uit.no> Subject: [R] Date conversion problem using

Dear users I have a problem with the dr function: "dimension reduction". I give you my example, and i'll be pleased to read your comments. #let be X a matrix 50*100: library(dr); X<- matrix(rnorm(50*100,5,1),50,100); #and let be Y a vector response: Y<- sample(0:1,50,replace=T); #I choose (for the expérience, but in reality i don't have it) a few variables #which are

Hi all, I've analyzed the packet stream resulting from a SIP session through Asterisk, and I found that Asterisk sends a lot of useless messages! In example, it repeats the three way handshake (invite-ok-ack) every time, also when it has just to forward the 'bye' message. Has anyone found the same srange thing? thanks..

Hello everyone, does anyone reported similar problems with ess 16.10-1 and R version 3.3.2 RC? These are the steps I do to reproduce it - Open any R script - C-c C-n (or C-c C-j) on any line to start the interactive session - the R process is started but it's hanging and emacs "freeze" (any command is ignored) R version 3.3.2 RC (2016-10-23 r71578) -- "Sincere Pumpkin

Hi dear group members, I have finally an Asterisk box working, capable of receiving and making calls. I have this issue while calling mobiles from our SIP softphones: -------------------------- linux*CLI> -- Executing NoOp("SIP/2000-6850", "3487024125") in new stack -- Executing Dial("SIP/2000-6850", "ZAP/g1/3487024125") in new stack -- Called

RH7.1, wine release 20010731. There is no windows installed on this computer and I try to install Word97 in 'no-windows' environment'. After I run 'wine setup' I see the initial word installation screen but after entering code numbers I see the following eror messge: Setup error 544: Setup is unamble to open data file E:\MSSETUP.T\~msstfof.t\Word97.stf. Run setup again from

Hi, i've downloaded/compiled/installed the bristuffed asterisk Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a and i'm using it with an hfc card. It runs on a debian 3.1 sarge machine with kernel 2.6.11. Asterisk works well if i configure the card using isdn4linux. I'm having problems dialing out (not tried the input yet). This is the output from asterisk: -- Accepting AUTHENTICATED call from

I'm having a problem with a new SIP trunk. Calls within the UK to fixed lines are fine, but calls to mobiles have noticeably poorer audio quality. I thought it might have been a codec issue; we have used G.726 for internal and external calls (over primary ISDN and GSM). So I tried allowing "alaw", (G.711 A-law) which is the native codec used within the PSTN in this country,

On 1.6.2.11-rc2 I've noticed a bunch of DEBUG statements on startup, such as: == Registered custom function 'SIP_HEADER' [Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot find variable 'SIPPEER' in tree 'description' == Registered custom function 'SIPPEER' [Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot find

I'm not sure if this is an Asterisk thing, a handset thing or a telco thing, so please be gentle with me if this is not the right place to ask ..... When placing a call over a SIP channel to a mobile phone, if the phone is engaged, it does not return a BUSY status straightaway. Rather, I get a ringing-out tone for the timeout duration specified in the Dial() statement; *then* I get

I haven't managed to work this one out yet, so any assistance appreciated ... We want to be able to set the outgoing caller-id on, BRI according to the extension but haven't worked how with asterisk ? we have several hundred inbound numbers on these BRI so we are able o use any one these to sett on outdial. One other point I have been told should work, bu have no way of trying... In

I've been reading the Linejack strikes again messages, and have another Newbie question is it possible to use a Voip Product as a Dialout line for * ? I have a Vegastream 100 Voip to PRI. box. With * can I use that as a Dialout / dialin box? The Vega100 does either sip or h.323. Thanks. Bradley Greep

Hi there, I?ve been out for some months now, haven?t been checking the list at all. Does anyone know if the problem with the Quicknet Linejack (FXO) card dial out to PSTN with asterisk was solved? Is anybody working on it? Cheers, -Z -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup

I need to make automatic voice calls from a Linux server, so when the system receive special "signals" it must use a wave (or .au) audio file, dial the number to call a person, and "speak" using the audio file. What can i use for this subject? I need a specific hardware device or a normal analogic voice-modem is ok? which software can i use (i need to invoke from a Perl

Hi, When I try to dialout over chan_capi everything works fine when I settle for msn=* in my capi.conf and use the primary msn of my ISDN-line. But trying to configure a different MSN the chan_capi doesn't dial and comes with: No one is available to answer at this time What can be the prob? -- Thanks, Marc aka IzNoGood

Hi, i install the Asterisk PBX on a linux machine with i4l to connect to PSTN (EuroISDN). And i configure a very simple dial plan in extension.conf. After this, i connect with a SIP program to asterisk and would call my cellular phone, but got this error: -- Executing Ringing("SIP/ACzerniak-0904", "") in new stack -- Executing Dial("SIP/ACzerniak-0904",

Dear all I read on the list back in 2003 that * does not support IXJ LineJACK dialout yet is this still the case? Thanks

If you set "dialout=fromvm" in your voicemail.conf, how do you then go about being able to dial back out? Is there a service feature code?

Hello! How can one select outgoing MSN when dialing out from ttyI-interfaces? I have successfully done this with CAPI e.g... exten => _XXXXX.,2,Dial,CAPI/60:bBYEXTENSION ...in extensions.conf. Currently correponding for my ISDN modem interface is... exten => _XXXXX.,2,Dial(Modem/g1:${EXTEN}) ...but this selects only MSN of outgoing group g1 for dialout MSN number. I also tried to

Specs FC2, Asterisk 1.0.0, Zaptel 1.0.0 TDM400P Port 1 FXS Port 4 FXO Standard analogue handset plugged in with pstn line. Problem: When I go to dialout it drops numbers on the outgoing number. Keys dialed from handset were 9 0418800185 I tried hitting the keys slowly as well as at my normal speed, all tones are heard in the handset for all numbers.