similar to: R 64bit for Windows on Windows 2012 server running Microsoft’s Hyper-V 2012?

Hi Guus, hi all, please find attached a proposed feature implementation for tinc. As mentioned in http://www.tinc-vpn.org/pipermail/tinc/2010-May/002344.html , my goal was to connect nodes on the same LAN using their local (LAN) endpoints. I've implemented a multicast sender, which announces its own endpoint on every connected interface regularly. All nodes receiving multicast

I'm trying to use odesolve for integrating various series of coupled 1st order differential equations (derived from a system of enzymatic catalysis and copied below, apologies for the excessively long set of parameters). The thing that confuses me is that, whilst I can run the function rk4: out <- rk4(y=y,times=times,func=func, parms=parms) and the results look not unreasonable:

Hi everyone, you can find the current version of my enhanced tinc using subversion: svn://tardyon.mon-clan.de/tinc I allowed anonymous read access, so feel free to download the sources. Unfortunately, my enhancements are based on a rather old git-checkout from Guus. The version should run under windows and Debian/Ubuntu. Best, Daniel -----Original Message----- From: folkert [mailto:folkert

Dear Guus, I've attached my first git commit to your repository. It does not contain any new functionalities, but it is a first try to interact with your git copy. Could you please verify, if you can push this commit to your repository? If it works, I'll send you the rest of my work, which contains: 1) some small improvements in logging (using flags instead of counters) 2) the

Hi, We want to restrict acces to the shares on our samba server using "hosts allow". Can I get this to work with clients who have dynamic IP addresses and don't have revers DNS lookup ? Best regards, Eric Eijkelboom Sr Systems Manager Medtronic B.V. Heerlen, The Netherlands Phone : +31-(0)45-566.8544 Fax : +31-(0)45-566.8008 www.medtronic.com <http://www.medtronic.com/>

Hi all, I am currently deploying tinc as an alternative to OpenVPN. My setup includes a lot of nodes and some of them are sitting together behind the same router on the same network segment. (E.g. connected to the same switch.) I noticed, that those nodes do never talk directly to each other via their private ip-addresses, but instead use the NATed address they got from the router.

Dear Samba Friends, I got the following error again and again: <snip> ..... Sep 4 10:58:18 sun smbd[94479]: [2006/09/04 10:58:18, 0] smbd/notify_fam.c:fam_check_reconnect(136) Sep 4 10:58:18 sun smbd[94479]: failed to connect to FAM service Sep 4 10:58:23 sun smbd[94479]: [2006/09/04 10:58:23, 0] smbd/notify_fam.c:fam_check_reconnect(136) Sep 4 10:58:23 sun smbd[94479]: failed to

Thanks to those who responded to my inquiry about generalized linear mixed models on R and S-plus. Before I summarize the software, I note that there are several ways of doing statistical inference for generalized linear mixed models: (1)Standard maximum likelihood estimation, computationally intensive due to intractable likelihood function (2) Penalized quasi likelihood or similar

We have the caller id with name option enabled with our provider, however, our polycom 501 phones will only display the number of the incoming call. Is there a way to see the callerid name from the cli when the call is coming in (like a print in the dial plan)? I'm not sure if the problem is with asterisk or our phones. I did turn on the calleridpres option in zapata, but I'm unsure what

Hello all, I was wondering if anyone has had much experience with Realtime Asterisk. I like the ability to setup my extensions and voicemail boxes in MySQL, but I have a huge worry. What if MySQL crashes. I played with rtcachefriends, but can't seem to find a way to have asterisk store the extension information to ensure the phones will continue to work even if MySQL has a hiccup. Any

So I might have a bit of a more narrow question from my earlier one. Previous, I had been wondering what would cause a phone dialing into a DID that connects to the asterisk box to get a fast busy. I've noticed the following message: chan_zap.c: Ring requested on unconfigured channel 0/1 span 2 Any idea what would give me this error? And would this cause a fast busy? Thanks again everyone

Hello all, I currently have all outgoing calls set to mask the caller id so it will always appear to be coming from our main number. The problem I'm having though, is with both the call detail in mysql and with the automon (recording) feature. It shows the originating number as the number I masked it to, rather than the actual person calling. How can I go about having both the destination see

Hi, One of my users is in trouble with his polycom phone hooked to an asterisk server. The phone works fine for a few days, and then disappears from the registered sip peers in asterisk. The user is able to place outbound phone calls, but can't receive incoming calls until the network plug is unplugged/plugged. Working line XXYYZZAA24/XXYYZZAA24 10.50.5.186 D A 5060

I'm running Asterisk 1.4.18 and having a problem with the clearglobalvars option. I have a NIGHT_SERVICE variable which I initially set equal to off. I then have an extension they can dial which will toggle that variable. My problem is when you enter the CLI and type "reload", it resets to "off" again. I've tried setting the clearglobalvars=no as well as just

Hopefully a fairly easy question for the group... I have a queue which should contain about 10 agents (it will be all the phones in the office). This office is remote, so I would like to add their sip phones into the queue remotely. Also, if the system ever gets reloaded or rebooted, I need those agents to remain in the queue. Question: 1) How do you remotely add agents to their respective

Hello all, I think Lee has given me a head start, but I'm still running in a circle (at least i'm in the lead). The problem is with my queues. The phones go to their own voicemail after 5 rings. That's about the same time I allow the phone to ring before trying another phone in the queue. Is there a way to tell asterisk....? If this call is coming from a queue, do not follow a

I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system

We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. So far, we've been able to get the phone to register with the asterisk server, and it can receive voice from the asterisk server (IE, voicemail, etc). However, asterisk can't hear anything from the softphone. We have used 2 different machines to test this on. We are watching

Hi, I am searching around this topic but didn't get any suitable link, can you guys please help with this. i have my linux based VM as .img which is built on top of VirtIO protocol and runs fine on KVM or qemu. i am trying to port the same on the Microsoft Hyper-V but it is not working. I get blank screen with no other actions, tried the same on virtual box also but result is same. i have

Not too sure what to think about this, considering M$'s track record with OSS and other competitors. Kind Regards, Keith Roberts ---------- Forwarded message ---------- To: opensuse-offtopic <opensuse-offtopic at opensuse.org> From: Basil Chupin <blchupin at iinet.net.au> Subject: [opensuse-offtopic] Microsoft struggles to get Hyper-V drivers in Linux kernel The