similar to: A few questions

Dear R People: I have the following in a file: resp factA factB 39.5 low B- 38.6 high B- 27.2 low B+ 24.6 high B+ 43.1 low B- 39.5 high B- 23.2 low B+ 24.2 high B+ 45.2 low B- 33.0 high B- 24.8 low B+ 22.2 high B+ and I construct the data frame: > collard.df <- read.table("collard.txt",header=TRUE) > collard.aov <- aov(resp~factA*factB,data=collard.df) >

Hello List, I'd like to ask how much you guys pay for an E1 (30 voice lines) and what company. You can email me personally and not the list. Best regards, Eddie _________________________________________________________________ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

Anyone have experience with these guys? If so, good, bad, average? http://www.junctionnetworks.com Thanks, Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/4222e3e7/attachment.htm

Hi I'm new to Dovecot so please be gentle with me. I did a search through the list archives but couldn't quite work out my problem. I have Dovecot installed and running and I can connect to it from my email clients (Outlook and Evolution) ... so looking good so far. I would like to setup three types (groups?) of mail store 1. A per user - e.g. Fred, Harry, Betty. (For my purposes they

I have a Cisco 7940G IP Phone. I am trying to load the firmware to SIP 3.2. The Phone just hangs in Defaulting CM to TFTP Server. It doesn't do anything else after that. I also have two other 7940g's that are the Universal Application Loader mode and say Protocol Application Invalid. I need to know what I can do to fix both these problems. I am running Asterisk@Home version 1.0. I have to

I am trying to build the latest release candidate and have run into a bit of a problem. When I run ./configure, I get: GlusterFS configure summary =========================== FUSE client : no Infiniband verbs : no epoll IO multiplex : yes Berkeley-DB : no libglusterfsclient : yes mod_glusterfs : no () argp-standalone : no I am going to need the gluster FUSE client now

I have ztdummy installed: Module Size Used by ztdummy 3464 0 zaptel 218756 1 ztdummy crc_ccitt 2176 1 zaptel ohci_hcd 16388 0 floppy 49028 0 pcspkr 2180 0 piix 8580 0 [permanent] ehci_hcd 24456 0 uhci_hcd 26256 0 rtc

Hello all, Here, we use rsync for years to share software tests distributions between from France to USA, Singapore and London, and all is working fine. But to save space we also need to compress files depending on their access time, and on purpose uncompress automatically before use. This makes rsync useless because it makes, of course re-transfer and deletions. when a new file is detected

Hi, FYI, Red Hat released an advisory today about a vulnerability in Ruby. So far it doesn't appear in the VuXML, but am I correct in presuming it will soon? https://rhn.redhat.com/errata/RHSA-2006-0604.html http://cve.mitre.org/cgi-bin/cvename.cgi?name=CVE-2006-3694 cheers, -- Joel Hatton -- Infrastructure Manager | Hotline: +61 7 3365 4417 AusCERT - Australia's national

Hi I'm browsing around the internet looking for signs that the IAX client library and app_meetme support video. I stumbled across this post by SteveK on the 27th of Feb 2006. "My company is looking to hire a full-time developer, who will be working about 25-50% of the time on iaxclient; in particular to finally integrate, build, polish and enhance video in iaxclient, add video

We have decided to allow our tech's to do support for non-clients of voicemeup.com You can head to http://support.voicemeup.com/ and one will be in touch 8 to 6pm business hours. 3 levels of support are offered for Asterisk/compiling Trixbox , Ivr's etc. -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 -------------- next part

Is anyone here familiar with configuring Cisco routers? I have a Cisco 3620 with 3x WIC-1DSU-T1, 1x 2FE-2W, and 1x 1E-2W. I have 2 T1 lines being brought in by ACD.NET, a local CLEC. Within the next 3 months we will be getting another line from them, as our bandwidth needs grow. I am new to cisco and need help configuring this router. I am currently on a 1M DSL line with them. The DSL modem is

I am trying to install R on a SLED 10.1 machine. R-base-2.7.0-7.1-i586.rpm fails with stas at linux-6b8s:~/RPMs> rpm -Uvh R-base-2.7.0-7.1.i586.rpm warning: R-base-2.7.0-7.1.i586.rpm: Header V3 DSA signature: NOKEY, key ID 14ec5930 error: Failed dependencies: libgfortran.so.1 is needed by R-base-2.7.0-7.1.i586 I tried to trick it into believing there's the library by setting up

I have a TDM02B to bring in two POTS lines for my incoming calls; I need to point each line to a different IVR... is there somewhere that can I can look to get this setup working? Basically, each line is for a different business. I know that for a DID the routing is simple but I'm not seeing where I can match up a DID with a Zap channel. I'm currently looking into the zapata.conf file

Hi. I'm evaluating GlusterFS for our DFS implementation, and wondered how it compares to KFS/CloudStore? These features here look especially nice ( http://kosmosfs.sourceforge.net/features.html). Any idea what of them exist in GlusterFS as well? Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm a newbie on Asterisk and I'd like to know if it's possible to connect two or more asterisk together. In fact, I'd like install and connect some asterisk together. Thanks for advance, Cyril _____________________________________________________________________________ D?couvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et

Hi, On behalf of one of our users, I installed R 1.8.1 on HP-UX11.0 by compiling the source using the gnu compilers and all was fine with make check On 1.9.0, I get an error from 'make check' when running reg-tests-1.R which gives error exit code 1. Commands used are: ./configure --prefix=/apps/global/Gnu_R/R-1.9.0 --with-x R_BROWSER=mozilla (all on one line, of course) make make

I have noticed that when I try to connect multiple G729 VoIP devices into a MeetMe conference that I can only add up to the number of G729 licenses I have. Now I would think that because all the devices are G729, this wouldn't be the case and the only license that would ever be used would be if a non G729 device or Zap channel was a part of the Meetme conference. This is apparently note the

I'm assuming asterisk does not have a SIP jitter buffer in place? Any ideas on how to help with this going over a data T1 where VoIP is shared with regular traffic? Problem is when people are downloading the voice is jittery, even lossy. Matt

Here's something I haven't been able to discover as of yet - I need to set up a "direct link" from my Asterisk box to an external line... basically I need to be able to pick up an internal extension and have it call a local phone number. This is call forwarding, I know - the question that I have is how do I set it up so that the extension always forwards. There will never be a