similar to: To Switch or not to Switch... that is the question....

Hello all, I'm looking for a way to be able to read a text file into R. It's a csv file but when I do "txt <-read.table("F00.csv", header=T, sep=",")" It doesn't read the file properly, and I only get 2 columns. If I open it up in OOc or Excel it open right with 7 columns. What I would really like to do is read the file as text and then split it and

Followed the "instructions" on voip-info.org regardinging fedora FC2, making Zaptel seems to work fine, however when I modprobe I get this. It looks like a version mismatch somehow. Ideas? If this ooc, sorry first post here :-) modprobe tor2 WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.358smp/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel

On Wednesday, 4 October 2017 14:56:23 CEST Richard W.M. Jones wrote: > If you have a struct containing ‘field’, eg: > > type t = { field : int } > > then previously to pattern-match on this type, eg. in function > parameters, you had to write: > > let f { field = field } = > (* ... use field ... *) > > In OCaml >= 3.12 it is possible to abbreviate

I installed Bind9 on a new ubuntu 13.04 server using apt-get install bind9 and am trying to integrate AD into it. Bind starts fine and will resolve my domain and computer names, but when I add the line include "/usr/local/samba/private/named.conf" into /etc/bind/named.conf, Bind9 fails to start. I have edited that file to ensure the correct line is included for Bind 9.9, and I am not

Can anybody tell me please, what isn't correct? What should I change in config? /var/samba/log/log.XP-CLIENT-IP: [2006/06/22 08:24:54, 1] smbd/sesssetup.c:reply_spnego_kerberos(303) Username DOMAIN\XP-CLIENT-NAME$ is invalid on this system all works in general, but the error above is listed in every /var/log/samba/log.CLIENT Thanks Steffen ---------------------------------------------

I've tried to figure out from the web site the minimum hardware cost to run a small office Asterix solution but I'm afraid I miss something: Let's say that I want to connect four/five analogic extension to the PBX. I have: - 1 computer as the server (with linux and Asterisk on it) - 1 dummy patch panel to connect all the analogic phones around the office What (and how many) cards

I like to have a continuously running R process, which can receive a dataframe from a client (over TCP/IP), does some processing, and sends some data back. What is the prefered way to do this? Using the socket interface? Using omega's CORBA stuff? Does anyone has example code for doing so? Thanks for any help Regards -- Dr. Jens Oehlschl?gel-Akiyoshi Analyse MD FACTORY GmbH Gr?nstr. 15

The problem I have had with many SMS email gateways, is that they do not allow to reply back. Of course, email is a superior service over SMS, but does not really replace it ( - at least not until all the mobile phones have email access on them...). It would be nice if the SMS service would be available not only on GSM/GPRS/etc., but also on any telephone number in the world - or at least on

Ok, is it me or do some of the commands just not work properly? I asked for mailboxstatus and got: Response: Success Message: Mailbox Status Mailbox: 1000 Waiting: 0 which is all well and good, except of course I have 2 messages waiting... which kinda means it only works, if you have 0 messages... (using voicemail not voicemail2) Andy

Friends, Last week, Hans Petter Selansky alerted us of a potential security issue in all releases of Asterisk. In fact, it doesn't involve the code, but the most common way to construct dialplans. If you have something like this in your Asterisk, you need to update your dialplans: [incoming-from-voip] exten => _X., 1, dial(SIP/${EXTEN}) Many VoIP protocols support a large character set,

Hi all, I'm a bit confused about how to use the switch statement. I've got an IAX2 link between 2 servers (SA & SB). I have use the switch statement to include extensions from SA onto SB which is happening perfectly. I've read that I can't use the switch statement the other way (ie SB->SA) at the same time. What is the correct way for both servers to know about each others

Anyone have some good examples if passing a call from one switch to another using IAX. I would like to have a call come in over PRI and pass it into a certain context of another server. Bother server seem to register with each other fine... I thought the command was... switch => IAX2/user:secret@host/context but I can't get it to work. Can anyone shed some light? Thanks, Chris

Ok, this has really freaked me out, but in a good way - sort of.. I've made no changes at all to my system, save messing with ADSI. However this has nothing to do with ADSI. The thing is all of a sudden my DECT phones have started reporting caller id, and not just the number, the name too! They have never done this before in the couple of months that I've had * running. I'm pleased

Thanks Andy ! that is I was looking for. It works fine. Angel. >Date: Fri, 11 Jun 2004 00:55:04 +0200 >From: "Andy Powell" <andy@beagles-den.demon.co.uk> >To: asterisk-users@lists.digium.com >Subject: Re: [Asterisk-Users] How to get the Called id >with AGI >Reply-To: asterisk-users@lists.digium.com >On 10/06/2004 at 14:40 Angel Diaz wrote: >Hi all,

Hello, * newbie here, I'm designing a setup that is to eventually be used in a production virtual PBX/VoIP service. Customers need to be able to change their setups over the web - I want them to be able to do simple things like setting up call forwarding, as well as more intricate stuff that will require me to re-generate their dialplans. Administration of the service is to be

I don't know if this is a problem with my cisco sip IP Phones or asterisk but I thought I would post here in case someone else has experienced this issue. When I make a call from my SIP cisco IP Phone to some remote IVRs I never get the rest of my soft keys, only the "End Call" soft key, and also DTMF doesn't work... its like the phone is acting like the remote end hasn't

Hi, I've just (earlier today) updated from CVS so that I can apply the dtmf caller id patches. Unfortunately this has had an undesired effect. I have an intertex ix66 which up until the CVS update allowed me to register my * server with the ix66 for my local domain (eg sip.mydomain.com). Now it appears that asterisk gets totally confused and tries to register with itself! Anyone got any

Hello! In my relentless quest for knowledge, I pose this question: who's got the biggest dialplans, and how big are these monsters? What's the biggest dialplan in use right now? If you feel you are a competitor, let me know how many contexts/extensions/priorities you are dealing with. Maybe the context with the most extensions, the extension with the most priorities would be

Call to Action! For those who have the courage/ability, go grab an SVN copy of the asterisk release, the HEAD version, and my latest patch, from: http://bugs.digium.com/view.php?id=6021 Right now, the latest version of the patch is 0.10. apply it to the SVN head version, and do a "make". Read the Wiki on AEL2: http://www.voip-info.org/wiki/view/Asterisk+AEL2 Look at the examples

I think I am the first to use the $135 Avaya 4602 SIP phone, but I need some support from the community to fix one problem I have with it. The phone stops working after about 20-30mins if I have mailbox=context in Asterisk; when I do have mailbox=contect in asterisk the sip debug returns "481 extension does not exist." Anyone willing to help me figure out why? I.E. Is it an Asterisk