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> On Apr 25, 2020, at 12:10 PM, Joerg Roedel <joro at 8bytes.org> wrote: > > ?On Sat, Apr 25, 2020 at 11:15:35AM -0700, Andy Lutomirski wrote: >> shift_ist is gross. What's it for? If it's not needed, I'd rather >> not use it, and I eventually want to get rid of it for #DB as well. > > The #VC handler needs to be able to nest, there is no way around

> On Apr 25, 2020, at 12:10 PM, Joerg Roedel <joro at 8bytes.org> wrote: > > ?On Sat, Apr 25, 2020 at 11:15:35AM -0700, Andy Lutomirski wrote: >> shift_ist is gross. What's it for? If it's not needed, I'd rather >> not use it, and I eventually want to get rid of it for #DB as well. > > The #VC handler needs to be able to nest, there is no way around

I have Asterisk running on a Linux 2.4.x box with ztdummy. Once I did a modprobe on ztdummy I was able to enter into a conference room using my softphone clients. I'm using SJphone and Firefly. I have noticed a significant delay (1 to 3 seconds) while talking within the conference room. I have tried both clients, SIP and IAX protocols and various codecs. I have also tried it from different

...t; > How are you dealing with this? Ultimately, I think that NMI will need > > to turn off IST before engaging in any funny business. Let me ponder > > this a bit. > > Right, I dealt with that by unconditionally shifting/unshifting the #VC IST entry > in do_nmi() (thanks to Davin Kaplan for the idea). It might cause > one of the IST stacks to be unused during nesting, but that is fine. The > stack memory for #VC is only allocated when SEV-ES is active (in an > SEV-ES VM). Blech. It probably works, but still, yuck. It's a bit sad that we seem to be growing mo...

...C > > Kaboom. > > How are you dealing with this? Ultimately, I think that NMI will need > to turn off IST before engaging in any funny business. Let me ponder > this a bit. Right, I dealt with that by unconditionally shifting/unshifting the #VC IST entry in do_nmi() (thanks to Davin Kaplan for the idea). It might cause one of the IST stacks to be unused during nesting, but that is fine. The stack memory for #VC is only allocated when SEV-ES is active (in an SEV-ES VM). Regards, Joerg

I previously had Asterisk 1.0.7 running on a Linux 2.4.x kernel with ztdummy. I was able to do things like meetme and music on hold. I recently installed Asterisk 1.0.9 on a different machine with a Linux 2.6.x kernel running ztdummy. I installed and configured everything the same way, but when I try to call into a conference room I get the error message stating, "that is not a valid

I have Asterisk running on my LAN with softphone clients (SJPhone) and Cisco 7940/60s, all using SIP. I also have a few remote sites connecting to my Asterisk server. I am getting an echo back of my voice when talking with one particular site. The caller does not hear an echo on their end. All calls on the LAN or to other sites do not produce an echo. When the caller places his SJPhone on

...ng with this? Ultimately, I think that NMI will need > > > to turn off IST before engaging in any funny business. Let me ponder > > > this a bit. > > > > Right, I dealt with that by unconditionally shifting/unshifting the #VC IST entry > > in do_nmi() (thanks to Davin Kaplan for the idea). It might cause > > one of the IST stacks to be unused during nesting, but that is fine. The > > stack memory for #VC is only allocated when SEV-ES is active (in an > > SEV-ES VM). > > Blech. It probably works, but still, yuck. It's a bit sad that...

In my testing, I''ve found the following error: zpool status -v pool: local state: ONLINE status: One or more devices has experienced an error resulting in data corruption. Applications may be affected. action: Restore the file in question if possible. Otherwise restore the entire pool from backup. see: http://www.sun.com/msg/ZFS-8000-8A scrub: none requested

Hello there, I have been working in a startup for a few weeks now, and I am responsible for setting up the production environment and for "strengthening" the product (a rails app). By the way I''ve started reading "Deploying rails" from pragprog. The deployment of the app will be done through a private beta with about 200 members. The startup currently owns a VPS

I am running Asterisk on our LAN with Cisco IP phones and softphone clients, SJPhone and Firefly. Everything on our LAN works fine and the quality is good. We have recently registered other callers not on our LAN who are using SJPhone and Firefly as well. The audio quality is fine, but I consistently hear a buzzing sound in the background. We have experimented with the line in and microphone

I currently have my Asterisk set up to "monitor" (record) all audio in my conference room on meetme. However, Asterisk will record an "____in.wav" and "_____out.wav" file for each user that joins the conference. Is there a way to set my extensions.conf file up so it only records when user when extension 1234 calls, for example? I'm assuming that the