similar to: iSCSI questions and VM Creation questions

Hi, After using a little bit oVirt, here's my thoughts: On the Network part : * In order to start the bridges, you have to reboot the node. It should be good to create the bridges "on the fly". * The VLAN configuration seems not to work (you can create it but it's impossible to assign it to an interface...). You can also assign this kind of interface on a opennet On the VM part:

Hi all, I've got some problems to make work oVirt. I've installed a Fedora 10 VM the lightest possible (nothing checked, even based) and installed after some packages (wget, sudo, acpid,...) and updated the system. By the way, acpid should be a dependency of ovirt-installer because the installation fails if it's not available. I've then installed ovirt (ovirt-server-installer

Hi all, I'm a totally noob on that point but on my test machine, I've made some changes: * Changes to make the ovirt node able to load igb * Changes to make the possibility to destroy an host (if it's disabled) * Changes to make work again (at least on the GUI part) the migration I think I'll should send "somewhere" those patches but : * How do I make the patches (with

Hi, Quick question (tried to find an answer but didn't found it): When I migrate a VM via oVirt, the VM is "stalled" at the end of the process (unreachable via Network, vnc show what I had before the migration but that doesn't move,...) and I have nothing in the logs of qemu. Is it something common (I had to disable selinux to make the process work) or not? Thanks in advance,

Hi guys, I'm trying to use Asterisk with LDAP integration. I created some schemas and it seems to work fine for sip.conf replacement. When I try to register a softphone to test the service, it seems ok from the softphone point of view (user registred) but when I do a "sip show peers", no one is registered (nor sip show subrscriptions, users...) I put my Asterisk on full debug and I

Add of a button destroy for disabled hosts. This button behave in a similar way than the delete button of a VM. Signed-off-by: Sylvain Desbureaux <sylvain.desbureaux at orange-ftgroup.com> --- src/app/controllers/host_controller.rb | 5 +++++ src/app/services/host_service.rb | 15 +++++++++++++++ src/app/views/host/show.rhtml | 17 +++++++++++++++++ 3 files changed,

Why pay 1.1 cent's a minute for interconnecting to another Asterisk server for a high volume call center. Do people really understand what they are trying to sale and take an honest look into what they advertise. As a high volume user like a call center I would not connect my Asterisk Box to there Asterisk Box to a third Sip provider who then hands of to the Level 3 and so forth. With LD

Hello dear R Users, I am working on a dataset of 928 Enterprises, of which are observed 12 different characters. I need to randomly sample, without repetition, 70% of the entreprises, to create a testing set, and let the other 30% of the enterprises be a validating set (holdout validation, I think that is). How do I do that? Of course all the characters of each row must remain together. Also, I

Hi all, Is there anywhere an API with the different method and descrption of the different JS of SCRIPTACULOUS ? Thanks for your good work in Prototype and Scriptaculous !! -- Cyril --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Spinoffs" group. To post to this group, send email to

Hi, As you may know, these ISDN BRI features are very important here in Europe as ISDN Basic Rate Access is very popular among Small & Medium Entreprises. I don't really know why but it seems that in many countries, default is to install small PBX using Point-to-Multipoint (PtMP) mode as opposed to Point-to-Point (PtP) which is the norm for PRI. So basically, in several countries, SME

Good morning everybody. My question is simple. Is there a way to perform relay register with Asterisk ? More precisely, I want my clients regiter to a Proxy Registrar (OpenSIPS/Kamailio) through my Asterisk : REGISTER REGISTER Client ------------> Asterisk ---------------> OpenSIPS So Asterisk keep a list of registered clients and only allows them to

I am using a standard Centos 5.4 setup, with kernel-xen-2.6.18-164.15.1.el5 and xen 3.1.2-164.11.1.el5. I have just succeed in migrating an existing XP to this setup and installed successfully gplpv 0.11.0.213. On an other 2003 VM, I am using 0.10.0.134 since several months without any problem. Looking at http://www.meadowcourt.org/downloads/, it seams there is a lot of version in that

Dear All, I get the following error when using either SASxport or Hmisc to read an .xpt file: > w <- read.xport("D:/consult/Trophos/dnp/base/TRO_ds_20081006.xpt") Erreur dans factor(x, f$value, f$label) : invalid labels; length 15 should be 1 or 14 > z<- sasxport.get("D:/consult/Trophos/dnp/base/TRO_ds_20081006.xpt") Erreur dans factor(x, f$value, f$label) :

Hi, I have a SIP request which comes from an Asterisk and which has to re-enter in the same Asterisk (during the same session), but during the second passage in Asterisk, it send me a 482 Loop Detected. So is it a bug or Asterisk control the session and considere it as a loop ? If it is not a bug, how could I resolve this problem ? Thanks Regards -- R?mi Druilhe

https://bugzilla.mindrot.org/show_bug.cgi?id=1581 Summary: Pb with syslog Product: Portable OpenSSH Version: 5.2p1 Platform: Sparc OS/Version: Solaris Status: NEW Severity: major Priority: P1 Component: sshd AssignedTo: unassigned-bugs at mindrot.org ReportedBy: eric.savidan at

Hi Community! If this issue was already topic, please excuse or delete my request... Topic 1 "no ringtone": I configured a SIP registration with my SIP provider (SIPCall). Everything works fine except the ring tone for the caller. The caller hears silence until the called party takes up the phone. I used the DIAL command with the r and R option but no luck... :( Has anybody the same

<!DOCTYPE html> <html><head> <meta charset="UTF-8"> </head><body><p>Hello everybody,<br><br>I manage not to detect one silence with record () when I make as follows:<br><br>Exten = > 0178900271, n, Record ($ ${ link_recorded_pseudos_clients } pseudo_ Client_Id} wav, 5,5) exten = > 0178900271, n, GotoIf ($ ["

Hello all, I'm jumping into this mailing list because my question still unsolved after Googling the web. Do you know any performance of sizing recommendations about Samba 4.7.1 ? For few informations : 2vCPUs / 1 Core 8Go of RAM 140 simultaneous Windows users / 250 at most 20 NFS shares mounted via fstab This question because my RAM is all eated up in the buffer/cache (normal according

Hi All, I'm running CentOS 5.3 x86, and recently my yum update isn't running as it should. It pops up with sub conflict messages for i386 packages, which I don't need - even though yum is downloading them. Due to this sub conflict messages, I'm not able to update/install my x86 packages, as the installation stops at the error message. Anybody, any idea how to come over

<!DOCTYPE html> <html><head> <meta charset="UTF-8"> </head><body><p>Hello list,</p><p><br></p><p>I am asking when a caller want to leave a message to a mailbox with the application voicemail</p><p>How i can limit the duration for exemple 30 seconds.</p><p>exten =>