similar to: mysql or postgresql?

This mail is probably spam. The original message has been altered so you can recognise or block similar unwanted mail in future. See http://www.silvervalley.k12.ca.us/index.php/195 for more details. This isn't spam? Please forward this message to postmaster@silvervalley.k12.ca.us with a request that the e-mail address you received this from be whitelisted. Content preview: Hi! Anyone know

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. - -S - -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -----BEGIN PGP

Dear R-users, I used to give commands such as: > source(file="~/path/to/file.R", chdir=TRUE) but with the latest v. 2.1.0 it does not seem to work anymore. I tried to figure out what it was going on and it seems that the string for which > class(file) [1] "character" is changed to > class(file) [1] "file" "connection" when the connection is

I copied the scaffolding edit and update and made it like this, Note: awnser is edit. Admin_contoller.rb def awnser @question = Question.find(params[:id]) end def update @question = Question.find(params[:id]) if @question.update_attributes(params[:question]) flash[:notice] = ''Question was successfully updated.'' redirect_to :action => ''list''

I have installed R 1.6 for Linux from source, but unfortunately, I didn't check if the Xwindow headers were available, so I have installed it without Xwindows capabilities. Thus, I want to uninstall it and install it again properly with the required features, but I don't know how to uninstall R when built from source. I would appreciate any help on this. Thanks very much in advance!

G'Day All, Can someone help me out please. My new CISCO 7960's manual says I have to setup a TFTP server. Googled it and got a little understanding, but from * standpoint, well I am still a lost. Can I set this tftp server on the same * box? Can in be on a WinXP box? Which tftp software would you recommend? Thanks much. BTY: Does anyone have a How-To on getting the 7960 fully configured

I would write in tab-text file a table like this one: TAB colname1 TAB colname2 TAB TAB... colnameN rowname1 # # # rowname2 # # # rownameM # # # then I wrote something like: mymatrix <- matrix(nrow=M,ncol=N,byrow=T) rownames(mymatrix) <- chvector1 colnames(mymatrix) <- chvector2

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! Are hints not yet implemented in res_jabber? I have this here: exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx at gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003 at internal : SCCP/6003 State:Unavailable Watchers 0 6002 at internal :

I just detected R and have, after browsing the manual, one question: I look for quite a lot of time for graphical software which allows to plot data from a table or external file _with the axes of the coordinate system changed_, i.e. the x-axis should run from top left to bottom left, and the y-axis from top-left to top-right. It is of no use to interchange the rows in the table, the

Hi folks, when I start asterisk directly, I get a colored CLI. When connect to a already running asterisk with asterisk -R, it's never colored, despite I'm running both from the same console (tty). Is there a way to force asterisk -R into color mode? Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have a general question about SIP access for nonregistered users. I would like to make it possible for basically anybody to make a SIP call to my asterisk without having to have a user account, but in a specific context. So that e.g. somebody could make a SIP call to SIP/stefan at my.asterix.pbx and it would go like this: [incoming_guest]

Hello, list a have a problem i can start asterisk, i get the fowlling error: [chan_capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Feb 11 13:50:36 NOTICE[2535]: chan_capi.c:2636 load_module: CAPI not installed! Feb 11 13:50:36 WARNING[2535]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Feb 11 13:50:36

Hi, I would like to implement an auto-redial function in a context. The idea is about like this: Dial a number Hear busy Hangup Pick up again Dial a code like *123 => jumps into a context which redials until callresult is not busy Maybe like this: [autoredial] exten => s,1,Set(number=${CHANNEL(lastdialed)}) exten => s,2,Dial(SIP/${number}@account,60,g) exten => s,3,Wait(15) exten

Hi folks, how could I play an announcement to the calling party as soon, as the called party picked up. I would like to deploy an asterisk in an environment, where a premium rate support-number is offered to customers which do not want to pay a monthly support contract. In Germany, you are commited by law to announce the cost per minute of a premium rate number at the beginning of the call. So,

Hi, I tried to connect cisco 7910 into asterisk system using chan_sccp.so. But I got a major issue : - when I called from 7910 to another sip phone in the same asterisk server, the call took place normally. - when I called from 7910 to another sip phone in different asterisk server, the call is answered but I cannot hear nor say anything. The phone just immediately lose its tone. - when I got

hi PSTN <--> [Teles ISDN / Asterisk] <--> SIP client When call is made through ISDN, no matter if taken from PSTN or Asterisk side, person in PSTN side can hear perfectly but in Asterisk side I only hear a very scrambled or very low quality voice, words repeated several times. Same is with echo test (call taken from PSTN) Setup: * Teles 16.3 ISA ISDN card with hisax kernel module *

I started working on a chat program but, I''m having this problem chatroom.rhtml <div id="chatroom"> </div> <%= periodically_call_remote( :update => "chatroom", :url => {:action => :chat }, :frequency => 2) %> chat.rhtml <h1><%= @chat.room %> room.</h1>

Hi, in case, anyone is interested... I have started compiling a blacklist of hosts and networks from which SIP fraud attempts occur. My criteria currently are: To block an IP: - Minimum 3 attacks within one week from the same IP To block a network: - Attacks from minimum 3 IPs from that network within 2 weeks Common criteria: - Provider does not react to complaints OR - Provider sends autoreply

Hi, Does anyone know how to make this thing (7940) work with asterisk (chan_sccp module) ? I've set the configuration according to the wiki and now the phone just keep asking for CTLSEP<xxx>.tlv from my tftp server. In the cisco's web interface, I found this in the device logs : 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 ...

I don't have a CCO account with cisco and I want to use asterisk for my cisco 7960 phones But I could not get myself a firmware image for it. If anyone can share a copy of it, would be most appreciated Or if you could point me to where I can download it Thank you ,jm -------------- next part -------------- An HTML attachment was scrubbed... URL: