similar to: creating a factor from dates by subject?

Hi, I have some data (lots of year,distance pairs), which I can straightforwardly boxplot: dists <- read.table("movedists.dat") with(dists,plot(as.factor(V1),V2)) If I want to plot these data as violin plots using vioplot, this approach doesn't work: > with(dists,vioplot(as.factor(V1),V2)) Error in Summary.factor(c(1L, 1L, 1L, 1L, 1L, 1L, 1L, 1L, 1L, 1L, 1L, : min not

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type registered for 'IAX'

Steve, Here is the config, I pulled from my server, that works with D'Link Phones: Main Menu -------------------------------------------------------------------------------- SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) ;bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) bindaddr = 67.109.153.236 disallow=all ;allow=ilbc allow=gsm allow=ulaw

Folks, This question is somewhat related to a previous posting of mine. I just can't seem to create a generic solution. Here is a function that I found searching around the internet: splitIt <- function(x, n) {split(x, sort(rank(x) %% n))} I use it like so: > splitIt(1:12, 2) $`0` [1] 1 2 3 4 5 6 $`1` [1] 7 8 9 10 11 12 Or > splitIt(1:12, 4) $`0` [1] 1 2 3 $`1` [1] 4 5 6

My analog card, uses a PCI slot and a 12V power connector, is configured with 2 FXO and 2 FXS modules. I can ring handsets connected to the FXS ports but I can't dial out from them. Is extensions.conf where I need to make changes? [root at robin asterisk]# cat chan_dahdi.conf [trunkgroups] [channels] [phone](!) usecallerid = yes hidecallerid = no callwaiting = yes usecallingpres = yes

I've got the same problem. MusicOnHold works if I use something like: Exten => 1111,1,MusicOnHold() but if I try to answer a call and then transfer or put on hold the call, I get no music. Does anyone have any idea? Bye, Gianluca. _____ Da: Kanishka Somaratne [mailto:kani@technoportal.biz] Inviato: gioved? 17 marzo 2005 5.53 A: asterisk-users@lists.digium.com

Hi: I am trying to setup a simple auto attendant with Asterisk using SIP extensions. I have an IP trunk to voicepulse and my outgoing calls go over that. I can also receive calls on that voicepulse trunk and want it to an auto attendant. Everything works except on the following: - one of the options is to allow the caller to press the extension that they would like to be connected to. I have

All, I have data that looks like below. For each id there may be more than one value per day. I want to select a random value for that day for that id. The end result would hopefully be a matrix with the id as rows, date as columns and populated by the random hab value. Thanks to someone on here (Jim) I know how to do the matrix, but now realize I need to randomly select some of my values.

Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: