similar to: problems on slackware

> he graphics card is emulated in Qemu, the graphics operations are > converted to VNC commands there, and you see on the VNC display what > would appear on the Windows VGA display. OK, Thanks. 2 other questions: 1. how do I connect tzo the VNC server for my domU OS? 2. Where is the QEmu code? In xend or in kernel? -jec Jean-Eric Cuendet Senior developer / Technical support

I'm using openssh 2.5.2p2 on Solaris-x86 2.6. I ran into a couple problems when I set UseLogin to "yes": The big one seems to have been reported before: login refuses to run without a utmpx entry. This problem appears to have been caused by the changes in revision 1.24 of session.c. Before this revision, the record_login() function was always called, no matter how UseLogin was

Attached is the latest version of my UseLogin patch that makes "UseLogin true" work on Solaris and UNICOS. As usual, I have provided configure.in changes that set the appropriate defines for Solaris, but I have not provided the configure.in changes for UNICOS (since they would be incomplete, and Wendy is working on this). This version fixes a problem with the last-login time always

Hi, A lot of the problems with openssh portability so far appear to be with the login record functionality, i.e. lastlog support, and variations on handling utmp vs. utmpx etc. Looking at for-profit SSH 1.2.27, login.c is rather embarassing spaghetti code, so laden with '#ifdef's it's almost impossible to read. OpenSSH's code isn't anything like that, but then it doesn't

I''m having troubles with my outbound VOIP connection. I''m convinced that I don''t have QOS/traffic shaping configured properly in my shorewall linux firewall, which serves as my Asterisk VOIP server and Internet router/gateway. I don''t have a separate router box. I''ve been using VOIP for about a year now, but just recently realized that I need to

I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000 for all my old analog phones. The analog phones with the Sipura seem to work great. Voice quality is fine on both ends on the Sipura. I'm using the Teliax service and I use the Ulaw codec for all phones. However, I'm struggling with the BudgeTone 100. On my end, I find there is lot's of voice cut outs. I'm told my

Many times the "speed" of an inbound voice call changes. It's similiar to playing a 33 LP at 45 speed. Sometimes the voice becomes uneligible. A "speed" change is the best way to describe it, seems like the voice packets are being played out too fast. Can anyone explain what might cause this? It doesn't always happen, and seem unpredictable. Thanks, Jim

on hp-ux 11 i see: $ date;ssh jenny Fri Oct 12 14:44:13 PDT 2001 Last successful login for stevesk: Fri Oct 12 10:45:42 PST8PDT 2001 on pts/2 Last unsuccessful login for stevesk: Mon Sep 24 22:55:53 PST8PDT 2001 Last login: Fri Oct 12 10:45:43 2001 from 172.31.1.53 You have mail. so solaris PAM is different. can other solaris+PAM users confirm this? On Fri, 12 Oct 2001, Benn Oshrin wrote:

http://bugzilla.mindrot.org/show_bug.cgi?id=585 Summary: sshd core dumping on IRIX 6.5.18 with VerifyReverseMapping enabled Product: Portable OpenSSH Version: -current Platform: MIPS OS/Version: IRIX Status: NEW Severity: major Priority: P2 Component: sshd AssignedTo:

Hi, I've gone and ported the latest version of openssh, 3.1p1, to GNU/Hurd. I've tried to learn from the other threads on this topic, but I still had to get rid of MAXHOSTNAMELEN where I could. James A. Morrison diff -urN openssh-3.1p1.old/Makefile.in openssh-3.1p1/Makefile.in --- openssh-3.1p1.old/Makefile.in Tue Feb 26 14:24:22 2002 +++ openssh-3.1p1/Makefile.in Tue Apr 9 16:16:49

Hi R-devel, I'm trying to get R 3.2.1 working on Haiku (an open source OS inspired by BeOS, not Linux based) on i586. With a few small changes to library paths and ifdefs I am able to get a seemingly working R binary. The build process stops with the 'tools' package. The last lines from make are below. Does anyone have any tips? I'm rather new to debugging at this low level. Are

I'm looking for suggestions for a good IP phone to use with Asterisk which can display CallerID information on the phone's LCD panel. I am currently using a Sipura 2000 for all my analog phones and I have purchased 1 bugdgetone 100 IP phone. I find the voice quality of the Sipura 2000 to be better than that of the budgetone. I find myself wanting to use the analog phones instead of the

I was having problems getting the UseLogin option to work on Solaris. I would recieve this error: No utmpx entry. You must exec "login" from the lowest level "shell". This led me to believe that Solaris login wants a utmpx entry in order to function. I put together a patch that calls record_login on Solaris when using the system login. I also noticed that writing a wtmpx

I've upgraded to asterisk-1.6.0-beta2. I'm trying to get the new Telco MWI detection function working. It doesn't appear to be working. I have this in zapata.conf ; PSTN connected here ;immediate=no ;busydetect=yes ;busycount=8 ;musiconhold=default mwimonitor=yes ;mwilevel=512 mwimonitornotify=/usr/local/sbin/zapnotify.sh faxdetect=incoming signalling=fxs_ks context=incoming

How do I get a context to continue to execute commands after the caller hangs up after a Dial command? I'm using the "e" option to the Dial application. I though the "e" option would allow the context to continue. This doesn't want to work for me. I'm using asterisk-1.6.beta5 I never get to "3" below. I get a message saying the "2" ended

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https://bugzilla.mindrot.org/show_bug.cgi?id=378 Damien Miller <djm at mindrot.org> changed: What |Removed |Added ---------------------------------------------------------------------------- Status|NEW |ASSIGNED CC| |djm at mindrot.org, |

I use Asterisk in my house. Each phone is a different extension. I really like the ability to have multiple simultaneous calls in the house. However, I do miss being able to be able to pick up a phone in a different room. Currently, I have to either transfer the call or transfer the call to a "conference" extension to move around the house. While a connection in progress on one

http://bugzilla.mindrot.org/show_bug.cgi?id=585 ------- Additional Comments From dtucker at zip.com.au 2003-07-07 00:32 ------- dmalloc (http://dmalloc.com/) claims to work on IRIX. It's likely to increase the CPU and memory load, though. I've built with dmalloc on Linux thusly: LDFLAGS=-ldmalloc ./configure && make eval `dmalloc -l /path/to/log high` ./sshd [options]

Hi, I know this request is a a little off-topic but may be interesting for dovecot users (at least i hope!) Does anybody has already had an experience with a calendar sharing solution including a webcalendar and an outlook connector, "compatible" with dovecot ? Today, there are many projects aiming to be a "M$ Exchange Killer" (such as Zimbra, OBM, open-exchange, etc) but