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On Mon, Nov 02, 2020 at 11:15:18PM +0000, Femi Adegoke wrote: > Hello Mr. Jones, > > Trying to convert a Fedora WS desktop to virtual. > The WS is installed on a Samsung 1TB m2 NVMe drive . > No local disk shows up so I can?t execute conversion. > > Does virt-p2v support NVMe? TBH I'm not sure. If you get a shell inside vi...

On Wed, Nov 04, 2020 at 08:01:49AM +0000, Femi Adegoke wrote: > Any idea how I can disable this message? It stops me from booting up the > virt-p2v iso. > > o/s is Fedora 33 WS. > > Probing EDD (edd=off to disable)... ok You mean this message specifically? It's a kernel message and shouldn't stop it from booting, but if it...

On Wed, Nov 04, 2020 at 08:36:12AM +0000, Femi Adegoke wrote: > > Is this how I would get to the kernel cmd line of the virt-p2v iso? > > Add edd=off to the first(!) kernel line inside the VM in /boot/grub/grub.conf The easiest thing is just to interrupt grub at boot (hit "Esc"), then edit the kernel command line there. This i...

On Wed, Nov 04, 2020 at 09:04:44AM +0000, Femi Adegoke wrote: > virt-p2v hangs & displays the edd message right after I select which kernel to > boot into. > At that point I have to power cycle the pc. > Yesterday (before I sent you email), it hung a few times but after the 5th > power cycle it went through. EDD is an ancient BIOS...

On Tue, Nov 10, 2020 at 11:53:19PM +0000, Femi Adegoke wrote: > Actually I suspect we need to modify virt-p2v to recognize > /dev/nvme* as a block device name. > > Hi Mr. Jones, > > Any approximate time frame on when the NVMe "mod" might happen? I guess when someone contributes a patch. Rich. -- Richard Jones, Virtuali...

Hi, I have asterisk 1.0.7 and spandsp-0.0.2_pre18. txfax return a non-zero return code only if the fax file is not found. Unfortunately I can't get any information, whether the fax was transmitted completely or not. Will an update to a newer version change this? Thanks for telling me your experience! Roger.

Hi all, Can someone share with me his experience in making asterisk-oh323 talk to quintum gateway without gatekeeper. My set up is QUINTUM GATEWAY ------IP----M ASTERISK (OH323) Both are gateways.. but I don't know what authentication I will set up in oh323.conf and how to set it up I will be glad if anyone can help Goksie

Kai, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice versa. I

I need your help I have a sangoma A104D on my dell server; I got card status ok with no alarm If I dialed the extension 6210006, it shows the output as stated below, but there is no ringing from the pstn number nor the iax softphone am using on my pc. I will be glad if someone can give me a working config? What I want to achieve is to send all my call to the pstn on A104D? The pstn am

Hi, To solve echo problems, I'm considering 2 alternatives. 1> Sangoma A104d - I can't find support for asterisk 1.2.1 2> Desktop echo canceller - http://www.oriontelecom.com/echo_canceller/desktop/e1_ec_desktop.html - I want to know where to buy and price. Any suggestion is appreciated. Thanks. Jason. p.s. : asterisk cli command "reload" can change rx_gain and

...provided by Easynews.com -- >> >> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > ------------------------------ Message: 5 Date: Thu, 2 Mar 2006 19:21:12 +0100 From: "ADEGOKE ARUNA" <goksie@gmail.com> Subject: RE: [Asterisk-Users] my zap channel not ringing & source from internal to telco line To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <440737b2.06b53a3d.4c69.f...

Hi all, Can somebody help me out, to get the call going through to my provider? I connected my A104D Sangoma card to E1/isdn and each I tried to make call I get the errors below. My protocol analyzer can see only setup info and release complete. WIRELESS2*CLI> set debug 9 Core debug was 0 and is now 9 WIRELESS2*CLI> pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 T203

Hi all, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice

Hello, Thank you for the job well-done. I installed the chan_h323 of the asterisk-1.2.7.1 and with lib pwlib-v1_10_0-src-tar.gz and openh323-v1_18_0-src-tar.gz and I used licensed g729 from digium. However, I am having a very funny behavour. 1. If I send a call on its ringing at the called side but the caller didn't get the ringing tone. 2. if the called picks up the phone, I am