similar to: Yet another disk I/O performance issue

Hi all, I'm trying to share a folder between a host and a guest using VirtFS. The source is NAS mounted on the host and QEMU runs as unprivileged user. The only way for me to use the folder inside the guest is to set the access mode to 'squash' but all file operations are performed by the user running QEMU. If I ran QEMU as root, the passthrough mode works as expected. Is there

On Sun, Jan 19, 2014 at 08:01:55PM +0100, Matteo Lanati wrote: > Hi all, > > I'm running a VM using libvirt+KVM and I have a disk performance issue. > > The host is the following: > 4 cores Intel Xeon 5140@2.33 GHz, 16 GB of RAM, SATA HDD, OS Debian Wheezy, > libvirt 0.9.12-11, QEMU-KVM 1.1.2+dfsg-2. > > The guest: > 1 CPU, 2 GB RAM running Debian 7.0, image in

Hi all, I use libvirt-go in my agent to attach rbd volumes. I ofen suffer the issue of incosistent of domain dump xml and domain persistent xml file. For example, when I try to attach volume, the the domain by ?virConnectListAllDomains? api from libvirt tell me the vdf or 0x0a is empty, but when attach it with the vdf device or 0x0a address, qemu return the error duplicated with vdf or PCI

I can't send mail to any addresses in nufone.net; they all get rejected by a spam blocker. And their website is gone, too!! The URL leads to a "parking site." My accounts still seem to work, but I wonder WTH is going on? Thx. B.

Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 <----------------------> g711

I was wondering what are the values for sending dmtf via sip info. I mean, when I use dtmf relay via sip info, the sip/sdp message contains a Signal=X where X is the dmtf. That's ok for dtmf 0-9 . but what when dtmf is * or # ? we must send signal=# ? I ask that because I noticed that budgetones phone sends out * as signal=10 and # as signal=11 . but asterisk don't detect them, 'cause

Hi. Has anyone experienced with the new firmware .77 ? There's Day Light Saving time now, but haven't time to play with it, till now. Matteo. -- Matteo Brancaleoni Espia System Administrator - IT services Website : http://www.espia.it Email : mbrancaleoni@espia.it

Hi list, is there a way to achieve in asterisk (version 1.4.x) the behavior described below? * a caller place a call to an extension, and I want the caller hears the extension ringing for some seconds, and then hears the music on hold (or a courtesy message) _in the same call;_ * the called extension must continue to ring until answered. With the m(...) option in the Dial

Hi, I'd like to implement some way of controlling remote SIP clients while in a call, to execute remote commands. The call topology (think of a PA system) is this: * the caller is in a MeetMe() conference room * the callees are Page()d, then the dynamic conference room is connected to the previous one I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the caller to the

Hi, I'm trying to write a small microsimulation in R: that is, I have a dataframe with info on N individuals for the base-year and I have to grow it dynamically for T periods: df = data.frame( id = 1:N, x =.... ) The most straightforward way to solve the problem that came to my mind is to create for every period a new dataframe: for(t in 1:T){ for(i in 1:N){ row = data.frame( id =

Hi, All Is there a provision for "AbsoluteTimeout" application to notify called and calling party of the reason why the call suddenly ended? This way, the parties will be much better informed, hence they will/should not think that their VOIP/telco provider(s) are providing bad service. Ta SJ

Hia .... i try to use the fonction Getvar of asterisk to get a variable myDNIS that i have define. i use it as follow Action: Getvar Channel: SIP... Variable: myDNIS but asterisk don't know it .i have the response as follow Response: Error Message: Invalid/unknown command does everybody meet this problem . i try all possible combination and nothing help please ..!! :-( thanks in advance

Hi. I was wondering if the zplex in the dev kit could be configured to have all fxs ports, instead of the standard 8 fxo + 16 fxs. If so, anyone managed to do that? Matteo. -- Matteo Brancaleoni Powered by RedHat Linux 8.0 Linux User #153521 -----BEGIN GEEK CODE BLOCK----- Version: 3.12 GS d? s:- a- C+++ UL++++ P+ L+++ E- W+++ N++ o K- w-- O- M-- V-- PS PE- Y PGP++ t 5 X+ R tv- b++ DI D+ G e

<P>hi,</P> <P>Can anybody pls tell me how to activate loggin CDR on mysql db. I tried&nbsp;editing the /etc/asterisk/cdr_mysql.conf file and recompiling asterisk, it didn't work. Normal loggin on Master.csv file works fine.</P> <P>Thank you inadvance,</P> <P>Surajee</P> <P>&nbsp;</P><br> --------------This mail sent

The subject pretty much says it all. I have a customer who would like to have an option where a caller can leave a voicemail in such a fashion that it would be simultaneously delivered to a set of mailboxes all at once--the idea is "trouble ticket" type operation where multiple technicians will *each* get the vm. He prefers that, if we can do it, to a "shared mailbox"

Hi, I have this error, I have a digium card TE110P Tiger3xx When I'm load the dirvers by this command modprobe wcte11xp I got this error "Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected FATAL: Error running install command for wct1xxp " when I'm Lunch the command ztcfg -vv I got this error.

Hi. I was wondering how can I make incoming calls to wait if the phone operator is busy. I've 8 incoming lines, with 30 extensions. What I need is if the operator is busy with call nr #1 , the new incoming call waits until the op. is free. Looking into app_queue seems the way to go. So I want to ask if I'm right or wrong: I set up only a queue , is to say operatorq, where the only member

Hi, I dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesn't matter whether I choose alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone

v2: - refactor logic to reduce code duplication - better functions naming Matteo Cafasso (6): filesystem_walk: fixed root inode listing daemon: refactor tsk code lib: rename tsk internal function New API: internal_find_inode New API: find_inode find_inode: added API tests daemon/tsk.c | 156 ++++++++++++++++++++++++++++++------------- generator/actions.ml

Dear R team I have a problem with the installation of the RCOR package; any time I try to install it I get the error message: Error: package 'ROCR' was built before R 2.10.0: please re-install it. What can I do to solve this problem? Thank you Kind regards Dr. Matteo Pettinari Senior Assistant Cardiale Heelkunde matteo.pettinari@uzleuven.be<mailto:matteo.pettinari@uzleuven.be>