similar to: Problems with recordings of call using Monitor

So i successfully upgraded my Debian vm''s to Debian 6 on XCP 1.0, but when i try to install a new clean one it always fails. Why is this? The VM won''t even start, while i used the Debian 6 squeeze template. Is this even possible? Thanks, Peter _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com

Hey, Is it possible to covert a .img to a xva? I have a virtual machine on a xen machine that i would like to run in a xen cloud. I used the xva.py and it gave me a xva + kernel i followed the copy instructions but i get this: [root@box ~]# xe vm-start vm=converted Error code: WARNING: /dev/xvda is not a disk image Error parameters: Traceback (most recent call last):, File

Hey, Is it possible to covert a .img to a xva? I have a virtual machine on a xen machine that i would like to run in a xen cloud. I used the xva.py and it gave me a xva + kernel i followed the copy instructions but i get this: [root@box ~]# xe vm-start vm=converted Error code: WARNING: /dev/xvda is not a disk image Error parameters: Traceback (most recent call last):, File

hello guys, i was wondering where and if i can find the xen kernel headers for the latest stable XCP (0.5) i need these because i have to compile a kernel module. Thanks! Peter _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Hi, I would like to use the from caller id, to allow calls yes or no. 101, and 111 should be allowed to use the Trunk, the rest of the phones are not. Is this even possible? So if the from caller id is 101 or 111, then allow the call, otherwise hangup. Thanks, Peter -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I was wondering if the following was possible: When somebody sends a fax to my direct number 0101234567105 (my extension will be 105) is it possible that Asterisk, or an addon sees this as a fax, and e-mail the fax to me? So everybody with a private extension will be able to receive faxes in his e-mailbox on his direct number. Any pointers would be highly appreciated! Thanks, Peter

Hi, I would like to create backups of running VM''s, is this possible? I found the backup, restore and update function in xsconsole but i can''t backup whole VM''s. I would like to do nightly back-ups of running VM''s on a diffrent server and i was working on a script for this, but i wonder if this is even possible! Thanks, Peter

Hi guys, I upgraded my home server to check out XCP 1.0 but i have a big problem. After the upgrade my network just died. I see the card, it''s listed as xenbr0 but there is no network acces/connection going over it. A ping to or from the server results in nothing. When i check out the configure managment interface settings i see the NIC, and it states connected. But there is no way to

I'm running Asterisk 1.6 and was wondering if anybody have a workig "barge in" solution running. I was thinking of using chanspy, but i would like that the original call would be dropped, and the new call would be the only one there. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I''m trying to follow the guide to set up Lenny 5.0 VMs on my newly installed XCP machine. Following what the wiki suggests I downloaded the lenny.xva from http://www.xen.org/files/XenCloud/Software/lenny.xva, saving to domain zero. (note that this file was updated March 1, 2011) When I try to import lenny.xva to my Xen Cloud Platform host using the command: xe vm-import

Hello, I''m running a XCP 0.5 at home, and on this system i''m running Asterisk. To configure Dahdi i need to have the kernel headers. I''ve installed the xen-0.5 tools to see some cool stats in my open xen manager, but now Dahdi isn''t working, i have to compile it for this kernel, but i can''t seem to find the headers for this kernel: 2.6.32-4-686-bigmem

I use model comparison with glms without mixed effects with anova(modelA,modelB), with mixed effects glm (glmmPQL), this doesn't work. Is there a way to compare model fits with glmmPQL's? Paula M. den Hartog Behavioural Biology Institute of Biology Leiden Leiden University [[alternative HTML version deleted]]

Hi Guys, I have a strong feeling the loads on my servers will be shooting up soon... anyone got any idea how many calls i can expect to put through a DL360: Dual Quad Core 2.33ghz 4gb RAM with 1gb allocated for a ramdisk (call recordings) This server is recording calls (mixmonitor), codec is gsm (no conversion). I know there's a lot of other things to consider like AGI scripts and such

Hello, I''ve read the FAQ and searched the mailing list but have been unable to resolve my issue with networking in DomUs. The following things I CAN do: +SSH from within DomU to Dom0 +SSH from within DomU to all other DomUs on the same machine +SSH to other machines from within Dom0 +SSH from within DomU to other machines +SSH from other machines to Dom0 (and use the xm commands to

Hi, I got the following output booting through PXElinux. I used a netboot floppy image for my ISA NE2000 (RTL8019) network card to download the pxelinux.0 file from my server. I use syslinux 3.07. Here's my output: PXELINUX 3.07 2005-01-12 Copyright (C) 1994-2005 H. Peter Anvin ######################################################## # Welcome to the Petteflet PXE booting infrastructure. #

Hello, I am looking for a gsm gateway that is SIP based i.e no need of FXO/FXS analogue connection. I searched the email archives and found messages from 2008 but not sure how accurate these are. What do you use and how well it works ? The only sensible one I found is one made by portech and one that is made by Eurodesign. The one from portech is like a trunk while the one from eurodesign

Hello, I am using OpenSIPS to register all the users and planning to use asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge. I have a scenario where the signaling does not happen properly: 1) A user from Opensips dials an extension 7000 which is an auto-attendant extension. The call is routed to asterisk to play the auto attendant messages like Welcome and Dial the

Hello Everyone! We wanted to let everyone coming to Astricon know that we will be holding an OpenSIPS Summit on Tuesday Oct 21st, 2014 at the Suncoast Casino & Spa. Suncoast is about 10 minutes away from Red Rock and we will be provide shuttle service to and from the Summit. For those of you that had to book at Suncoast it should be really easy to find us! Here are some things you can

I have the following dialplan: ; calls prefix by '8' are recorded exten = _8[01]./_251,1,Set(something=shortened) exten = _8[01]./_251,n,Set(WAV=filename) exten = _8[01]./_251,n,Monitor(wav,${WAV},mb) exten = _8[01]./_251,n,Dial(mISDN/2/${EXTEN:1},,g) exten = _8[01]./_251,n,System(send-recorded-conversation ${WAV}.wav ${EXTEN:1} emailaddr) exten = _8[01]./_251,n,Hangup() The idea is that

I have a problem where SIP calls from some providers are dropping at 15 minutes. The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS sends calls to an Asterisk server. Below, 'Client' is the IP address of the client's host (running FPBX-2.8.1(1.8.20.0) 'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls 'Asterisk' is the IP