similar to: Congrats to CentOS@Home Folding Team

All, The CentOS Folding at Home team has cracked the top 10% of all the folding teams. We could use some more members that have CPU Cycles to spare :) Folding at Home is a great distributed computing program that is used to process items for medical research teams. It is similar to SETI at Home (if you are familiar with that). Here is info on Folding AT Home: http://folding.stanford.edu/

I see the team has passed the 150 ranking. Now at 145 of 58272 total teams! -- MailScanner is like deodorant... You hope everybody uses it, and you notice quickly if they don't!!!!

Congratulations to all members of the team for contributing towards breaking through the 2,000 place barrier. In the overall team rankings we are now in position 1885! Considering how 'young' the team is in terms of how long its been going thats a stupendous achievement! Well done all! From now on we can watch our teams ranking on

Congratulations to Bill Venables and Brian Ripley for their latest "tour de force" - MASS 4, which just arrived in my mailbox last week. As always, the book contains new pearls of wisdom to improve my statistical and programming skills. Kudos also to Peter Dalgaard for his "Introduction to Statistics with R". It is a delightful volume that I can recommend to all those

Centos now has its own Folding at Home team! If you would like to learn more about what we do, please follow this <a href="http://folding.stanford.edu/">link</a> to the Folding at Home homepage. New to distributed computing?, its not a problem. Here is the simplest description: Many computers working on separate pieces of the same puzzle are faster than a few computers

Does anyone know how to get CID working in AEL2 ? In extensions.conf you can do: exten => 111/666,1,PlayBack(demo-congrats) exten => 111/666,2,Hangup() exten => 111,1,PlayBack(demo-moreinfo) exten => 111,2,Hangup() and if callerid 666 dialed 111, they would get demo-congrats, everyone else gets demo-moreinfo. In AEL: 111 => { Playback(demo-moreinfo);

Congratulations to the CentOS folding at home team for breaking through the rank 400 barrier. The next target is the rank 250 :) More information about the team can be found at http://www.sharons.org.uk/cf.html, and we welcome new members and their machines. We always need more members and more machines too :) Sharon. -- 18:47:57 up 8 days, 20:56, 2 users, load average: 1.30, 1.76, 1.61 A

I see team Centos will soon be approaching the #150 spot. the_dobc -- MailScanner is like deodorant... You hope everybody uses it, and you notice quickly if they don't!!!!

Hi, I have cvs updated all my modules (zapata, libpri, zaptel and asterisk). I have also read in the archives & seems that no-one has run into this problem. What I'm trying to do is simple. Just make and outbound call using the /var/spool/asterisk/outgoing directory. I copied /usr/src/asterisk/sample.call and only changed the context & extension. I configured my Zap1 to the same

Hi! I just installed Asterisk 0.4.0 with all the default options, and the configuration samples it has. When I try to dial from an h323 client (gnomemeeting) I get this message on the messages file: Jun 19 11:48:45 WARNING[15375]: File file.c, Line 410 (ast_openstream): File demo-congrats does not exist in any format Jun 19 11:48:45 WARNING[15375]: File file.c, Line 553 (ast_streamfile):

actually, i do have a workaround which bypasses the exec command entirely: system("asterisk -r -x 'add extension s,3,Playback(demo-congrats) into local'"); but it's ugly. seems like it should be possible to do this with exec. .t ---------- Forwarded message ---------- Date: Sun, 23 Nov 2003 21:17:50 -0500 (EST) From: tad <tad@media.mit.edu> To:

Is anybody hosting with GoDaddy.com? I''m about to set up a hosted account for my RoR app and see that they''re now offering RoR hosting. I''d really appreciate hearing from anybody who''s tried them for RoR. Any congrats / complaints / etc. you had / are having would be really appreciated! Thanks, Bill -------------- next part -------------- An HTML attachment

as i said, right now i'm just getting my feet wet. but, i will be needing to build dialplans on the fly. 'add extension' seems like the right call to make. .t > What is the goal of this? It doesn't make much sense to me. Care to > share some insite into what your goal is? > > bkw > > On Sun, 23 Nov 2003, tad wrote: > > > actually, i do have a

I am recieving inbound calls to my asterisk box over IAX. And getting "spawn extension....exited non-zero" errors. Im not entirely sure what this means, and would appreciate any help in fixing my problem. FYI: ********** is the inbound phone number x.x.x.x is a remote asterisk box calling my own asterisk box. When I choose it to dial an internal extension using this dialplan: exten

hi folks. (apologies in advance if this is a particularly stupid question) just getting my feet wet with asterisk / agi, and am a little stuck using EXEC. it works fine for applicaitons that take simple arguments, but chokes on applications that require multiple words as arguments. for example, this works fine: EXEC Playback(demo-congrats) but this doesn't: EXEC add extension

My first Asterisk install: Debian sarge with the 2.6 kernel, and two X-Lite soft-phones. I followed the online how-to documents and was calling between the two soft-phones and calling the demo system with no problems and had full audio. I then went on to configure the TDM400P's two FXS modules. I got into that a ways and was having some success, but no dial-tone when I was off the

Hi, the first thing I did with Asterisk is listening to `demo-congrats' by Xlite on the same machine. This works perfectly. The config files are those shipped with the package. Now I want to listen to it over ISDN/Capi but I don't succeed. My `capi.conf' is like show in many tutorial on the web. In `extensions.conf' I just added the following lines: [capi-in] exten =>

I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards. Each "group" of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback "demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.

Dear all: a quick question: I am used to apply rm(list=()) regularly to remove all old codes in preventing them creeping in current analysis.however, with that application, functions I wrote are also removed. please let me know how to keep the thing you want while remove those you don't. thank you best yong

in sip.conf [general] port = 5060 ; The TCP/IP port for SIP communiations bindaddr = 0.0.0.0 ; Address to bind to. 0.0.0.0 all addresses on server. context=other ; Default for incoming calls disallow=all allow=ulaw allow=gsm in extensions.conf [general] static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here