similar to: check join Linux (SAMBA) to Domain controller Win2003!

The CUA spec says that Ctrl-home and Ctrl-end move the caret to the start/end of the buffer (possibly on a different page). Shouldn't Wine's EC do the same?

On Mon, 20 Feb 2023 15:58:33 +0000, Stadler Thomas <thomas.stadler at vpbank.com> wrote: > as the download method 'wininet' is deprecated, I'm looking into alternative ways to install packages from within R. > Unfortunately, curl, libcurl and wget refuse to cooperate with Kerberos on our corporate setup. > When exactly will the 'wininet' method stop working? R

On Tue, 21 Feb 2023, Selke, Gisbert W. writes: > On Mon, 20 Feb 2023 15:58:33 +0000, Stadler Thomas <thomas.stadler at vpbank.com> wrote: > >> as the download method 'wininet' is deprecated, I'm looking into alternative ways to install packages from within R. >> Unfortunately, curl, libcurl and wget refuse to cooperate with Kerberos on our corporate setup.

dear Hackers, i have a voicetronix Openswitch card, and i have been finding it very dificult to get it to work with asterisk. i intend to connect 8 ports to the PSTN and 4 as station ports. problem 1: On running asterisk all i get at first i get : event[9=>[11] station OFF hook] on vpb/1-12 even [12=>[11] loop drop on vpb/1-12 event [12=>[11] Tone detect:GRUNT event [2=>[11] Dial

Hi we've got Asterisk CVS-HEAD 18-Aug-04 (modified by Voicetronix as available on their site for use with the vpb driver) and an OpenLine4 (4xFXO). The same server also has two X100P. Calls on the Voicetronix card drop instantly when the called party picks up. The vpb driver reports that it detected a hangup (loop drop) yet there is no hangup when connecting the X100Ps or analog phones to

Hello, Hacker I install VoiceTronix OpenSwitch 12 port PCI Telephone Card, and setting vpb.conf, extensions.conf following My problem is: When i dial to fxo(channel 9-12), it is ok, but when i continue press exten 102, the channel crach with error messages following exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872 Do i ignore some setting for VoiceTronix OpenSwitch12

I am a serious Asterisk newbie: just installed asterisk last week and it is now running with our Voicetronix OpenLine4 hardware. All is working as expected with one exception, in the following sequence (extracted from my extensions.conf file): [GetConfirmation] exten => s,n,SetVar(TimeOut=0) ; if timeout and TimeOut=1 then user already timed out once, so hangup exten =>

Hi, I am fairly new to asterisk. I am currently testing my first setup. I've been able to debug most of the problems to make asterisk work with my hardware setup until this time. Currently I have the following issue: Voicemail is running but when I test to leave a voicemail thru my incoming PSTN channel (voicetronix / vpb), asterisk will not detect sound (according to the log) on that

Hi, I saw this setup on the web .. is this looks right to you ? The security set to user level, which use linux samba local login and password. Why this needs to set workgroup = SAMBAGRP ? [global] workgroup = SAMBAGRP security = user passdb backend = tdbsam Thanks

Hello all, We had an experimental system which works on OpenLine4 telephony card and Asterisk 1.0.9. Customer asked to upgrade Asterisk to 1.4, then we found our problem: At first Asterisk 1.4 does not compile chan_vpb.so. The problem is it tries to compile chan_vpb.cpp to chan_vpb.o and chan_vpb.oo, then try to link them together. I manually compiled it, then make went smoothly. But it

Hello everybody,, This time i want to replicate PDC to BDC when there's is any changes on PDC, here is my conf. on /etc/openldap/slapd.conf LDAP Server master moduleload syncprov overlay syncprov syncprov-checkpoint 100 10 syncprov-sessionlog 100 LDAP Server mirror moduleload syncprov syncrepl rid=001 provider=ldap://ldap.domain.com:389 bindmethod=simple

I am working on upgrading our current squirrelmail and replace uw-imap with dovecot.? I have what looks like a successful installation.? My current setup is: Centos 5.2 sendmail 8.13.8 dovecot 1.1.17 (imap) squirrelmail 1.4.17 I can send and receive email fine.? However when I try to delete an email in my Inbox, it doesn't get deleted.? It only sends a copy to the Trash folder.? But the

Hi, how can we copy numeric and/or text data to the clipboard ? (under windows2000) It works fine with graphics and I need to do it with text data . thanks, regards Vincent Demolombe -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send "info", "help", or

Good day all I had a look at the extensions.conf sorting http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting What I'm trying to do is route all my cellphone number threw a channel and all other calls threw the other 3 channels Cellphone numbers are 10 number,i.o.w XXXXXXXXXX. This is what I tried but it doesn't seem to work,please help Thanks Altus extensions.conf

So I've got a Voicetronix card and it looks like the kernel driver works. Other than the 0's for ID info. vpb: Driver Version = 4.0 vpb: major = 251 vpb: tmp [0xfc8fec00] dev->res3 [0xfc8fec00] vpb: tmp [0xfc8c0000] dev->res2 [0xfc8c0000] vpb: 1WS Write cycle vpb: Manufactured 00/00/0000 vpb: Card version 00.00 vpb: Serial number 00000000 vpb: Setting up udev... vpb:1 V4PCI's

Hi, I am using a voicetronix OpenLine4. I downloaded a recent asterisk CVS from voicetronix webpage but have had no luck to reduce echo on outgoing calls and for it not to crach on incoming calls. I dont think both problems are related though. Here is an output of what happens when a new call comes in and my voicetronix tries to pick it up and crashes asterisk: > vpb/1-1: Event

Hi list, anyone know what is going on here? I don't get any sound from the out clip & I get the following when I dial in to asterisk (after which it just times out): -- Event [0=>[06] Ring] on vpb/1-7 -- Executing Wait("vpb/1-7", "2") in new stack Read_channel ## vpb/1-7: Setting record mode, bridge = 0 -- Executing Answer("vpb/1-7",

Hi list, I am having some difficulty implementing a certain dialplan where the following happens. If the first Dial() is not answered, I want to play a small greeting then ask the caller to either hold the line (try calling again) or press 1 to leave voicemail. exten => s,1,Dial(${BLAH},10,Tt) ; Dial 10 sec exten => s,2,Answer exten => s,3,Playback(greeting)

When dialling in and dialling my extension, when answered I get " Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[20499]: File chan_sip.c, Line 1111 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3) -- Hungup on vpb/1-3 complete --

Hi, I hope some can help me out, as I have a problem with the vpb driver in *, and I am hoping someone has hit these issues and can offer some advice. First off, I'm connected to a British Telecom SystemX exhange with the local Tel Co. I can ring out fine, but when the callee on POTS answers, the vpb driver detects a loop drop and immediantly disconnects the call. I have currently set