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...e code so that it could be a real multi-processor tool. In other words... Will the R code adapt itself to the new machine (Unix with al least 4 processors)? Or shall I change the code to have multiple real threads and transform the algorithms into parallel computing strategies? Thanks in advance, Aitor. --------------------------------- LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y moviles desde 1 centimo por minuto. [[alternative HTML version deleted]]

Hi, I''m new to Rails and was trying out the tutorial that they give on the apple website http://developer.apple.com/tools/rubyonrails.html I installed everything including lighttpd, and I got up to the point where I want to create my first record. When I hit the ''create'' button, lighttpd quits. If I have the server running as a foreground process I get this message

I new in asterisk world so, please, forgive me if I say something stupid. At least, and after a lot of tryes, the isdn card seems to be registered: [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, PRI Signalling signalling -- Registered channel 2, PRI Signalling signalling -- Automatically generated pseudo

I have a samba server configured at a clients office and sometimes when he is drilling down into directories in his "File Explorer" it stalls. I tail'ed the log files and I ran tcpdump, but I can't see anything that stands out. What should I look at? I am running Samba Version 2.2.3a-12.3 for Debian. brian -- Brian Lavender http://www.brie.com/brian/

Hello! I would like to know which hardware I need, to use asterisk with up to 20 analog lines. Also I woul like to know if there is any card that suport both analog and isdn lines, and if there is any way to make the analog phones now I'm using work with asterisk. Thanks

Im trying to configure voicemail, but asterisk doesnt respond to dtmf codes. I uses kphone with g711u codec (I've tryed the others one) and in sip.conf I configure dtmfmode=rfc2833 (I've tryied inband and info). Asterisk seems not to "see" the tones. Could somebody help me? Thanks