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...ge, ob serielle Daten auch ??ber R eingelesen werden k??nnen oder ob das R-Programm in ein Labview-Programm eingebaut werden kann. Ich w??re sehr froh, wenn Sie mir in dieser Sache behilflich sein k??nnten. Ebenso m??chte ich gerne N??heres ??ber R erfahren. Zur Verf??gung steht mir bisher das bekannte B??chlein ??ber R und eine Beschreibung "Message Boxes in R TclTk". Mit freundlichen Gr??ssen Dr. Oscar Oehler Phys.Dept. ETHZ HPF D12 8093 Z??rich Tel 3 21 65

Hi, I have the problem that WaitExten is not responding to key presses. Here are the sections from my extensions.conf: [globals] incoming_call=0 menu=0 announce=0 [internal] exten => 777,1,Goto(hotline,${EXTEN},1) [hotline] exten => _X.,1,Set(CALLERID(name)=Hotline) exten => _X.,n,Set(original_extension=${EXTEN}) exten => _X.,n,GotoIf($[${announce}=1]?4:10) exten =>

Using 'Samba version 2.2.1a' Please could someone offer advice in order to configure the following. I need domain level authentication for most shares, however 1 share needs to be made available to a none domained NT client, such that perhaps share or user level security would be appropriate. How can this be configured on a single samba server. Thanks in advance to anyone who responds

Dear Sirs, I?m writting a pam module for Radius authentication, which should allow a remote login via ssh on a Linux machine with an openssh server. In particular, the user which is configured at the remote Radius server is not present on the local user database of the Linux machine. Unfortunatly, openssh will not allow to start a PAM user authentication if cannot find the user login in the

hi, we are using a voip h323 switch. the switch sends all caals to our Gatekeeper (gnugk). gnugk musst send all calls to asterisk and asterisk must do his choice (sip endpoint or out to PSTN) Making calls to our h323 switch works fine over asterisk. what must i configure to get inboung h323 calls from our gnugk to asterisk? any hints for me? thx -- Thomas K?pper 01063 Telecom GmbH &

Hi List, we are using openh323 gatekeeper for voip telefony. We also have a voip over ss7 TELES Switch for voip into POSTN Network. Know we want to use Asterisk for converting SIP to h323. Now my question. Is Asterisk an full h323 gatekeeper like openh323? Do we need openh323 GK for astrerisk, too?. And how can i tell asterisk to sent all none SIP-ip calls to the gatekeeper over h323? thx

with a h323 client over my gatekepper a call comes over asrerisk to my sip endpoint: == Spawn extension (sip-phones, 01634255122, 1) exited non-zero on 'SIP/0699073201-528d' -- Executing Dial("H323/ip$10.0.0.124:49638/18690", "SIP/0699073201") in new stack -- Called 0699073201 -- SIP/0699073201-dc61 is ringing -- SIP/0699073201-dc61 answered

Hi ! I have a small problem. When switching a call (pstn -> sip user), I get the sip phone ringing - ie. everything is OK, but I do not get a ringtone in the handset on the pstn side. Can anyone help me out in how to make * play tones ? My setup: E1 IP pstn ------ Asterisk ------ sip phone Regards, Dave

Sorry if this is posted to the wrong forum but as it is related to a problem I have with Asterisk it may just scrape through!! I am running Fedora 1 and I can telnet in to my asterisk box as any user except root and am using the same credentials as logging in locally. I am new to Linux and any help would be gratefully appreciated. Thanks Neil -------------- next part -------------- An

Hi all, I have found this: time smbclient -L //debian -U% Sharename Type Comment --------- ---- ------- print$ Disk Printer Drivers software Disk IPC$ IPC IPC Service (Samba 4.7.7-Debian) HP_F4100 Printer HP Deskjet F4100 series MX870-series Printer Canon MX870 series l6p Printer l6p MX435

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