search for: kulle

hi, i am not able to extract the p-values from the Manova function in the car library. I need to use this function in a high-throughput setting and somehow need the p-values produced. Any ideas? Best regards Bettina Kulle Andreassen -- Bettina Kulle Andreassen University of Oslo Department of Biostatistics and Institute for Epi-Gen (Faculty Division Ahus) tel: +47 22851193 +47 67963923

Does anybody have any experience with the ZyXEL Prestige 2000W? I am having problems with the line tear down when I call another extension. If nobody picks up at the other end when I hangup the 2000W, the other extension continues to ring. Is there any way to hangup a SIP call if there is no more traffic? Asterisk seems to think that there is still a connection open. This is pretty annoying

Dear all: I ust installed Fedora16Fedora16 X86_64 on my new machine, and then I ran the commands as follows(ran as root): yum install qemu-kvm yum install qemu yum install libvirt yum install virt-manager then I started the libvirtd as follow: /etc/init.d/lbvirtd start So , it ran well. However, after a few days , when I ran the command "virt-manager" , it showed as in the

...maxp = maxp) 3: knnimp(x, k, maxmiss = rowmax, maxp = maxp) 4: impute.knn(dummy0, k) Possible actions: 1: abort (with core dump, if enabled) 2: normal R exit 3: exit R without saving workspace 4: exit R saving workspace ################## thanks for your help in advance! tina -- Bettina Kulle Andreassen University of Oslo Department of Biostatistics and Institute for Epi-Gen (Faculty Division Ahus) tel: +47 22851193 +47 67963923

Anybody ever managed to implement a solution where one could forward a voicemail from one * server to another? Dominique

Hi , is it possible to calculate ld-measures D, D', r and perhaps corresponding p-values with r IF THE PHASE IS KNOWN? The genetics - package provides the LD function only for ambigious phase. Thank you very much Bettina Kulle

I've just seen this post: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg41132.html and it took me back to play again with my dust collecting 2000W. Does anybody got DTMF to work? My sip.conf looks like this: [400] type=friend context=from-sip username=400 secret=verysecret disallow=all allow=g729 dtmfmode=rfc2833 host=dynamic nat=yes qualify=300 canreinvite=no My phone is

_Description_ We are looking for an expert LAMP (Linux, Apache, MySQL, Perl, and PHP) developer with some Asterisk experience who is based in Western or Eastern Europe or Asia. We can work with an individual or an organization. You must be fluent in English. We need you to help expand development of Signate's core software products. As part of the Signate development team, you will

I have multiple voiceage g729 licenses installed on a RH9 box, and have a remote C7960 configured to use it (low bandwidth). In calls like: Remote C7960 -> g729 -> asterisk -> g711 -> C7960 the audio is oftentimes rather choppy. Changing the remote 7960 to use g711 seems to eliminate/reduce the choppyness. Any ideas on what might be behind this?

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Anyone have any experience with either of these, I 'd appreciate some feedback? Plus it seems pretty easy to steal a connection with this. Zyxel Prestige 2000W WiSIP thanks, - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin

Hy guys, I cannot understand How the asterisk works. I would like know how the h323.conf, sip.conf and extension.conf works. I don't understand the parameters and the [sections]. What I need to the asterisk get a SIP call and forward them to a H323 terminal. I working at the h323.conf and extension.conf but I cannot understand!!! Please someone can help me. I your can send me a example (with

All, I just had the fortune to take one of the new Senao Wifi SIP phones for a short test drive. First look - it's a nice, compact phone. Weighs around 87g and roughly the size of a Nokia 6210. More on the those later. The thing that struck me was the RF power, it's rated at 100mw (20dBm). That's 10 times more than any of the other brands out on the market Cisco, WiSIP, Zyxel

> "If" I understood your initial objective correctly (and I may not have), > the user's phones are negotiating the codec to be used for each rtp session. > > Asterisk parameters can be used to dictate rtp sessions between the sip > phone and asterisk, but that won't influence the next step in which the sip > phone negotiates a new rtp session directly with the

Hi all I have just got a P2000w and experience several problems. Hopefully there is someone out there that has got it working. I saw it on Cebit and the person demonstrating it there told me that it was connected to an Asterisk server on the stand -so it should work. Problem 1: it does not register correctly It get lots of messages like this: Jun 26 19:45:19 NOTICE[1107585968]: chan_sip.c:5630