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Hello, i only got a small problem. i try to create automatic new dataframes, or png?s. the main problem i got is: how can i create automatic a new name for a file (read out by simply "for") - i tried to use "(paste...) but theres an errormessage, about a wrong declination. R told it is as.character, but need as.Real. Should i use another method than "paste"? i tried as

(english version see below) Sie erhalten in den naechsten Minuten eine digital unterschriebene E-Mail von einem Freemail-Anwender. Damit Ihr E-Mail-Programm den Ausweis ueberpruefen kann, muss das "Root-Zertifikat" von WEB.DE installiert sein. Klicken Sie dazu bitte auf: http://trust.web.de/root.sql Auf dieser Web-Seite finden Sie eine genaue Anleitung zur Installation des Zertifikats

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

Hello Ben Lindstrom, I tried your package creation scripts for Solaris which are contributed with the new OpenSSH Version 3.0.0p1. Nice that you develop these scripts but I run into a major problem installing the package which was created by the script. On our Solaris hosts /usr/local is only a symbolic link to /opt/local. After installing the OpenSSH package the symbolic link was removed and a

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

Hello I just installed openssh-3.7.1p1 on IRIX64 6.5 04100802 IP27. The compilation failed since the header file openbsd-compat/inet_ntoa.h was missing. I downloaded the package from different sources including ftp.openbsd.org, there was no difference. Since inet_ntoa.c has not been changed, copying the inet_ntoa.h from openssh-3.6p1 fixed the problem. The package should be fixed anyway.

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, our system is as follows: Linux Server (Debian woody), Samba 2.0.7, several Windows clients (NT 4 Server and Workstation, Win95), Tektronix 840, Laserjet 4000 and Deskjet 2000. As printing system we use ESP Print Pro 4.04 (which seems to be comparable to cups 1.0x). In smb.conf we included "printing=cups" and the following printer section. [printers] comment =

Hello! In the Samba network! How can I install Samba 2.2.8a? I have two different SuSE Distributions SuSE 7.2 professional and SuSE 8.1 professional The following files are in the Folder "/home/samba" For SuSE 8.1 professional libsmbclient-2.2.8a-34.i586.rpm libsmbclient-devel-2.2.8a-34.i586.rpm samba-2.2.8a-34.i586.rpm samba-client-2.2.8a-34.i586.rpm samba-doc-2.2.8a-34.i586.rpm

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

I was just told that for my forthcoming system I will be getting a data T-1 instead of a voice T-1. Given that all of the handsets will be voip phones, no analog at all, do I need echo cancellation? I looked at the voip-info wiki and it seems to me that the answer should be "no" but I would like to confirm that. TIA, W

Terry Jones wrote: > The documentation is somewhat silent on this subject. If you mean https://wiki.dovecot.org/Tools/Doveadm/Sync the answers seem implicit to what's been stated. > What permissions does the SSH user need ? To be able to run the doveadm executable (or a wrapper script that eventually runs doveadm) on the remote side. > How associated does it need to be with

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. I enabled policies with pdbedit. Password must be changed every 90 days and must contain at least 8 characters. I enabled password history too. After that (I tried it in samba 3.4.3 and 3.0.25 with same behaviour) every time a user try to log in the domain using Windows receives a "Your password expires today. Do you want to change it now ?" message box. If the password is changed,

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