similar to: sshd bug?

What is the best way to benchmark disk IO? I'm looking to move one of my servers, which is rather IO intense. But not without first benchmarking the current and new disk array, To make sure this isn't a full waste of time. thanks

Hi guys, I''m trying to setup my first puppet installation and I''m running into problems. I''ve got the puppetmaster running on a fresh install of Centos 5.4. The puppet client is also on another Centos 5.4 server. Once I''ve signed the client certificate on the puppetmaster, the client seems to break. See below for the debug output from the client. Any clue as

I saw several threads requesting an Asterisk forum to complement the email list. i.e. http://lists.digium.com/pipermail/asterisk-dev/2004-February/003103.html I recently created an Asterisk forum within TMC's popular VoIP forums for everyone to use. http://voip-forum.tmcnet.com/voip-forum/forum/forum_topics.asp?FID=15

Is it possible to convert a VMWare image to KVM? As I have been building a few test machines on Vmware Fusion and would like to migrate some to a KVM server. Thanks in advance. Matt Keating Linux System Admin ? Dennis Interactive 30 Cleveland St, London, W1T 4JD Tel: 020 7907 6823 (direct line)? Fax: 020 7907 6600 (fax)? P Please consider the environment before printing this e-mail NOTE: The

Im setting up a very basic rails app and have a question about the needed migrations & associations. Basically, my app is an occasion reminder service that emails users when occasions that they select or input are coming up. Occasions will be selected from a checkbox type list or alternatively manually input by the user (name and date of the occasion). Users can have_many occassions.

I've got a problem w/ zap calls being dropped right after they are answered. I have a log file: http://pastebin.com/368526 Everything looks OK except for the DEBUG[25563] chan_zap.c: Exception on 9, channel 1 that seems to come up quite often. As soon as the other end of the Zap answers (my cell phone), and I can even hear a half second of noise, the line goes dead and gets hungup. In

I've got an issue w/ 1.2.0beta1, where if I call VoiceMailMain from a sip phone, and then either put in incorrect passwords or just hang up, I never get a Spawn Extension that hangs up the call, and my sip phone is not capable of making any more calls until I restart the daemon. Can anybody help me fix this? -- Jesse Keating GameHouse -- Systems Engineer

When I hang up in voicemail, Asterisk seems to stop responding. (hangup vs pressing # to disconnect). After that, no calls can be made until I restart Asterisk. In IRC, a developer seemed to think it had to do with me using switch => in my dial plan. Basically I never see the calling extension get the -1 signal. Can somebody help me figure out why this is happening and how I can fix it

I was at first, but then removed it. The results were still insufficiently fast. > Were you using the -c option of rsync? It sounds like you > were and it's > extremely slow. I knew somebody who once went to > extraordinary lengths to > avoid the overhead of -c, making a big patch to rsync to > cache checksums, > when all he had to do was not use -c.

Hi folks, New to the list. Just updated the bootrom and app firmware on a Soundpoint IP 501 as per: http://www.voip-info.org/wiki-Polycom+Phones Updated from: to: APP 1.4.1.0040 1.5.2.0054 BootROM 2.6.1.0003 2.6.2.0032 After I did this, it appears that the Web interface for the phone won't change the settings, nor will it actually reboot the phone now. What do I

Can someone please explain why every time that I call an action in my controller, every model that is stored in the session needs to be interpreted by a database query? If I have four models stored in the session, I end up with five long "SELECT a.attname, format_type(a.atttypid, a.atttypmod), d.adsrc, a.attnotnull FROM pg_attribute a LEFT JOIN pg_attrdef d....." queries when *ANY*

Hi Folks, OK - I'm not entirely new to Linux but I'm certainly not great on the boot process. What I'm trying to achieve is to boot from an external USB HDD (not flash drive) using EXTLINUX. I've run an installation against the disk using a VM. The disk was partitioned as follows: /dev/sda1 = /boot (bootable) /dev/sda2 =/swap /dev/sda3=/ (i.e. root) /dev/sda4=/dos sda1 and 3

I know Vonage doesn't officially have a "bring your own device" type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060329/5bc9f644/attachment.htm

Hello, I've got a boot image that is of size 1722, so to put it on floppy, I use dd if=boot.img of=/dev/fd0u1722 but if I try to run syslinux on this floppy afterward, I get: [root at yoda root]# syslinux /dev/fd0u1722 /dev/fd0u1722: Sector sizes other than 512 not supported Is there some way around this? Can we use syslinux on this type of floppy disk? Lilo works, but we would really

I'm testing out some IP501 phones and I ran into an issue. WHen I try to add a new contact into the directory, I am not able to. A window blinks really fast but the entry isn't saved. When you exit the Contact Directory system you get a 'Busy! Please try again' window. What the heck could be going on? -- Jesse Keating GameHouse -- Systems Engineer

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Hi, We are running a samba fileserver, access controlled using posix acl (right 770, with users/groups on the filesystem level. Therefore samba shares look like this: [share] path = /srv/academic read only = no writable = yes create mask = 0770 directory mask = 0770 Now certain users complain that they cannot access certain folders, but looking at the folders from the linux fileystem, their

Hola, compañeros. Pido ayuda con algo que sé que tiene que ser simple, pero la presión de tener que sacarmelo de encima me simplifica a mí demasiado y no me doy cuenta. Tengo una matriz de datos en la que tengo características tipo factor, necesito trasponer esa información a una matriz de datos binarios, en función de algunas combinaciones de esas variables. Unas combinaciones tienen que pasar a

Hi, I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now having a problem with Originate and chan_local. I'm using the following Manager API action to originate a call: Action: originate Priority: 1 Context: trunk Callerid: 100 Channel: Local/100 at callback/n Exten: 123456789 Variable: USERFIELD=127.0.0.1|USEREXT=123456789 WaitTime: 30 This is intended to first call

Juan, I am running a Calling Card application on a Dell PowerEdge 2850 with Asterisk 1.0.7. Recording conversations I have seen on my server causes the processors to burn more than necessary so I would recommend what William from Signate recommended: " Consider saving recorded calls in a database on a separate server. It will be simpler to build a retrieval interface that does not