similar to: HVM vLAPIC timer interrupts intermittently disappearing

check before relinquishing vlapic because VMX domain may crash very early. Signed-off-by: Xin Li <xin.b.li@intel.com> Also pls push to xen-3.0-testing.hg. thanks diff -r dc8122d90670 xen/arch/x86/vmx.c --- a/xen/arch/x86/vmx.c Wed Dec 14 18:47:16 2005 +++ b/xen/arch/x86/vmx.c Thu Dec 15 16:18:14 2005 @@ -113,9 +113,10 @@ if ( active_ac_timer(&v->arch.arch_vmx.hlt_timer)

x86:vlapic: Fix possible guest tick losing after save/restore Guest vcpu may totally lose all ticks if the vlapic->pt.irq was not restored during save/restore process. Fix it. Signed-off-by: Wei Gang <gang.wei@intel.com> diff -r 0892f5a96736 xen/arch/x86/hvm/vlapic.c --- a/xen/arch/x86/hvm/vlapic.c Fri Dec 10 15:19:51 2010 +0800 +++ b/xen/arch/x86/hvm/vlapic.c Fri Dec 10 15:27:11 2010

Svm patch to add 32bit msr support (combined both 32bit with 64bit functions) and enable vlapic for 64bit. Applies cleanly to 9023. Please apply. Signed-off-by: Tom Woller <thomas.woller@amd.com> _______________________________________________ Xen-devel mailing list Xen-devel@lists.xensource.com http://lists.xensource.com/xen-devel

I'm attempting to connect to a Metaswitch, inbound only (at this time). The Metaswitch is the only "connection" (at this time). All I'm getting so far is a bunch of "OPTION" messages which my Asterisk box replies to but I don't get inbound calls. Here's my sip.conf. As you can see I've been trying a bunch of different options without success :(

Hi, After c/s 18694 changed vHPET to vpt, for single HVM RHEL 5u1 guest idle case, our box will consume ~0.8W more power than before. The reason is two periodical vpts'' expires are hard to be aligned in the 50us soft timer SLOP. So we are considering a vpt specific enhancement which could try to just align periodical timers within vpt. A generic enhancement is to add a new interface

I'm (still) trying to get my Asterisk box talking to a Metaswitch. All I'm getting is a "heartbeat" of OPTIONS messages coming from the Metaswitch which my Asterisk box replies to. The exchange looks like: <-- SIP read from 172.b.c.d:5060: OPTIONS sip:metaswitch@206.b.c.d:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP

I am setting up VM off Metaswitch due to a problem with Metaswitch VM. I have a couple days to prove this works and I need a little assist please. I am using TRIXBOX 2.6.2.5 and have the Meta SIP trunk up. I have extensions built that can talk to each other. I took a trace on the TRIXBOX that shows when I dial my test phone on Metaswitch it goes to VM after a couple rings and the call goes to my

Hi folks, I am having a small problem with asterisk-1.6.2.7 + app_fax on OpenBSD 4.7 -release. Everything seems to work fine. I have a macro which answers, receives the fax to a tiff, and then runs a script (mailfax) to convert that to pdf and email it. It all works perfectly except for some errors I am seeing in the console. After it hangs up I get a dozen or so messages in the cli

Im trying to get my asterisk box to register to a sip provider without much success. here is my console output in asterisk Dec 13 12:57:17 NOTICE[213005]: chan_sip.c:3982 sip_reg_timeout: Registration for 'voicemail.nexband.com@metaswitch.nexband.com' timed out, trying again -- Got SIP response 403 "From: URI not recognized" back from 208.149.73.5 Urgent handler in my

I've been racking my brain for the last two days to try to figure out what I could possibly be doing wrong in my configuration for a SIP trunk that's setup through my local ISPs Metaswitch. I've setup a very simple SIP Peer, which I've played around with a lot in the past two days but still comes back to the following basic setup: [provider-fireball] type=friend

Hi All, I'm stumped on a weird problem. I have an * server working fine for local SIP phones and IAX2 connections. We just provisioned a second Ethernet port to attach to a local SIP provider. PSTN calls incoming work fine: PSTN -> SIP Provider -> SIP -> * but outgoing calls are not. Call setup takes place and the caller can hear about 1-2 seconds of audio before the SIP provider

I'm trying to send CallerID info to a MetaSwitch system over a PRI. The MetaSwitch gets the info exactly as it is sent by Asterisk, but I think it might be having trouble with the Hexadecimal b1 that is being sent just before the first character of the CallerID Name. Does anyone know what the significance is of the b1 being sent here? Or, is there a way to make Asterisk not send the b1

Greetings, I have a problem getting Asterisk registered as a UAC against the MetaSwitch call agent, because the customer insists on running it on a NAT'd box. Thus, the Contact: field in the REGISTER request betrays the private IP address of the Asterisk box, but the source IP of the message is a public one. Most registrars don't have a problem with this, including Asterisk. However,

I am having difficulties with customer-bound DTMF being very short & clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE Asterisk is running Asterisk 10.4.0 on a

When I make an outbound call I hear a half of a ring and than silence until the call opens up. It seems asterisk is sending a 183 after the 180 message. My CPE device does not support multiple 18x messages in the same call setup. When we receive the 180 we present ring back to the phone, but when we receive the 183 we get confused and stop the ring back tone, but do not open up the early media

Hello, as a variation of our issues with Adhearsion calls dropping when an INVITE comes in for a bridged call, I now have a new issue to contend with. Our call is in an AsyncAGI application, and has been bridged to another channel. The provider that supplies the DID sends a polling reINVITE every 15 minutes (it's a documented Metaswitch behavior amongst others). The reINVITE is seen as a new

I recently configured a SIP peer which i must specify my fromuser as my ten digit "DID". I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is "asterisk". Is this a bug? Or is there some other config I must make ? register = 2155551941:123456 at 10.0.138.226/2155551941~600 [peer](!) type=peer context=inbound qualify=yes

I'm using FreePBX/Trixbox with Asterisk 1.4.17-1 trying to register against a Sylantro server in front of a Metaswitch. I'm able to register and receive inbound calls but outbound calls are rejected by the far end. The username and password have been checked repeatedly. Putting the same authentication and server IP into a softphone or polycom phone work fine for inbound and outbound

Hello-- Looking for a solution. I'm using asterisk HEAD version, from a day or two ago. Trying to register with a Metaswitch voip server via sip. They gave me a userid, and a password. I plug it into a register command in sip.conf: register => 3074449999:pword@isp [isp] realm=voip.isp.net auth=3074449999#c491b58f6fd6da12691fa0de86fbbcc3@voip.isp.net type=peer context=workline

Hello Everyone, Calls that are private name private number have the following TO header: From: "Unavailable" <sip:asterisk at server.com>;tag=as120a1079. Don't tell anyone, but we are trying to put on a "We're big enough to own the pricey softswitch" look. Even though I would pick a OpenSIPS + Asterisk combo over a Metaswitch any day. Three words "Service