similar to: linux, XP, and samba

I need to create a legend for a simple scatter plot in the following format. This is Blah1 number1 number2 This is Blah2 number3 number4 . . . This is Blah6 number11 number12 I looked up these help pages and found the following solution. lStr<-c(Blah1, Blah2,....Blah6, number 1, number2, ...number12) legend(x="topright",lStr,ncol=3) So this creates the tabular format I am

rly there would be a heavy penalty to launching a shell so you would want t= o carefully evaluate the frequency this is executed on your system.<br /> <div class=3D"container"> <div class=3D"line number1 index0 alt2"><code class=3D"sql plain"><br /> DELIMITER @@</code></div> <div class=3D"line number2 index1

I am new to Rails and Ruby, and to OO languages, and seem to be making a very silly mistake somewhere here. Can anybody help? I am trying to write an application which involves ''exercises'', each of which consists of several ''templates''. This is based on MySQL tables with these names. I have models and controllers, built with the Rails Scaffold, for both

Hi All I am looking for a way to prefix a constant value to all the rows in column in a data frame. Eg. V1 2 3 4 5 I want to make it like this V1 number2 number3 number4 number5 Thanks! -Abhi [[alternative HTML version deleted]]

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 asterisk-users-bounces at lists.digium.com wrote on 12/14/2017 09:36:06 AM: > From: "basti" <mailinglist at unix-solution.de> > To: asterisk-users at lists.digium.com > Date: 12/14/2017 09:36 AM > Subject: Re: [asterisk-users] Rewrite Outgoing Number > Sent by: asterisk-users-bounces at

Hello, I am new on asterisk and do some tests on freepbx. I have 2 SIP provider: Provider1: In-/Out- Flatrate, only 1 Number Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 numbers On Asterisk site i have 3 phones (branch ??, don't know how its called in asterisk) Is it possible to do something like: Phone 1: Incoming Call: Number1/Provider1 Outgoing Call:

On Tue, Apr 14, 2015 at 09:38:22AM +0200, Daniel Heckl wrote: > Sebastian, > > Your code sounds good, I'm curious how it goes on. > > First the linux machine had the Google Public DNS 8.8.8.8 as DNS > server. After I changed it to the via PPPoE assigned DNS servers, i > had no changes any more. But we should be prepared for changes. > > You must enable the dnsmgr.

Hey everyone, I have noticed a few questions close to the issue I am having but I haven't seen any that quite match the problem I am seeing. I have 3 queues. Some members share one queue and some are completely separate. Some members have a higher penalty then others. I am using addqueuememeber and removequeuemember for the login and log out and I verify members with their password for

On Tue, Apr 14, 2015 at 08:26:07AM +0200, Sebastian Kemper wrote: > On Thu, Apr 02, 2015 at 11:33:38PM +0200, Daniel Heckl wrote: > > I do not want set allowguest=yes. The problem is, there is no official > > list with ip addresses of Telekom Germany. But I think all ip > > addresses comes from the ip range 217.0.0.0/13. > > Hello Daniel, > > Judging by the lists

I have Asterisk 1.2.13 installed as a Debian package and I edit only sip.conf and extensions.conf in this way: sip.conf: [general] realm=work.com.ar ; Realm for digest authentication bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes

Hi, I have two X100P cards in the same system. I can use both of them to initiate and/or receive PSTN calls. I want now to define separate context for each of them, in oder to route inbound calls to different extensions. This is what I have now in zapata.conf file: [channels] language=en context=inbound-analog signalling=fxs_ks usecallerid=yes callwaiting=yes echocancel=yes

Hi people, I want to make a meetme between 2 numbers. First I enter the number1 into the meetme. It is waiting for the other number, but the other number never entered, so, how can I finish the meetme from the dialplan?. Is it posible by using MeetmeAdmin and kick all the users? Thanks, Anahi Ludue?a _________________________________________________________________ Descubre

Hi, I am having trouble passing variables via the call files, here is my call file via the php: fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Events: off\r\n"); fputs($oSocket, "Username: $strUser\r\n"); fputs($oSocket, "Secret: $strSecret\r\n\r\n"); fputs($oSocket, "Action: originate\r\n"); fputs($oSocket,

Hello, I am calculating 'Betweenness' of a large network using R. Currently, I have the node-node information (City1-City2) in an excel file, present in two columns where column A has City1 and column B has City2 that city1 is connected to. These are the steps that I go through to calculate betweenness of my network. a) Convert the City1-City2 (text) into Number1-Number2 in the excel

Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16

Hey everyone, I have a more of an opinion question then a technical question. The asterisk server I am setting up is going to host 3 different businesses. Each business is in the same building, and on the same network. My question is regarding calls coming in and going out. We are a small ISP and have a lot of numbers that are forwarded to our phone system. The other companies have about 3

Hi Folks, I want to use key() to position the legend on top of the page in the following example. Prefer not to use locator()> par(mfrow = c(2,2))> plot(1:4, title="Plot Number1")> plot(rnorm(10), title="Plot Number2")> plot(1:10, title="Plot Number3")> key(x,y, text=list(c("Alpha","Beta")), text=list(c(1,2)))> > > x,y

Hi guys, I have a customer with asterisk registering 100 lines from my Voip Provider. In some times during a day we receive this messages: [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request

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]

I think your last approach should work (the m3u file), except you specified port 8002 instead of 8000... which would cause the behavior you're getting, winamp opens but does not play the song. Fred -----Original Message----- From: icecast-bounces@xiph.org [mailto:icecast-bounces@xiph.org] On Behalf Of Wolfgang Schwurack Sent: Friday, June 03, 2005 10:18 AM To: icecast Subject: [Icecast]