similar to: Open Source VOIP at Toronto Conference

Hey all, The Toronto AUG has been working with Clue.ca and IT360 (LinuxWorld/NetworkWorld), and has put together a mini-asterisk conference within their larger conference: http://www.it360.ca/asterisk.cfm If you're interested, as an 'association' we get 25% off the listed prices. Our dicount code is: "A101", and our association name is: "Asterisk User Group".

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

BLAST! LLVM mailing list headers are still royally screwed up... My message is below... On Sun, Aug 30, 2009 at 2:20 PM, Talin<viridia at gmail.com> wrote: > Hey all, it's been a while since I have posted on this list, but I've > been continuing my work with LLVM and getting lots done :) > > One question I wanted to ask is whether anyone had any advice on how to >

Dear R list, I calculated a two-sided p values according to 2*(1-pnorm(8.104474)), which gives 4.440892e-16. However, it appears to be 5.30E-16 by a colleague and 5.2974E-16 from SAS. I tried to get around with mvtnorm package but it turns out to be using pnorm for univariate case. I should have missed some earlier discussions, but for the moment is there any short answer for a higher

I agree, any platform suffers when it is extremely difficult to implement. What we need is an interface that does everything we need and shows what asterisk is capable of, a lot of features will go unused because you might not know the exist unless you hunt them down in the source or conf files. I trained on an Avaya INDeX switch it had a complex console but was laid out in a structured way a

Hello Everybody, I am looking for a way to trigger puppet rum remotely. so that i don''t need to login into individual nodes and perform the puppet agent --test from there. Is there any alternative to puppet kick in puppet 3.0, since its being deprecated in telly. Also , can anybody tell how to configure foreman 1 to trigger puppet run from the gui in puppet

http://www.networkworld.com/news/2008/120608-fbi-criminals-auto-dialing-with-hacked.html?Inform=nl&netht=rn_120808&nladname=120808dailynewsamal Criminals are taking advantage of a bug in the Asterisk Internet telephony system that lets them pump out thousands of scam phone calls in an hour, the U.S. Federal Bureau of Investigation warned Friday. The FBI didn't say which versions

A lot of us wouldn't be here without him. DEC made good, really reliable hardware. mark <http://www.networkworld.com/news/2011/020711-kenneth-olsen-dec-obit.html>

Well, not to take away too much from the tinderbox, but I'd like to point everyone's attention to: http://www.networkworld.com/community/blog/microsofts-open-source-love-expands-centos-li Headline: "Microsoft's open source love-in expands with CentOS Linux support" Short version: Microsoft now supports CentOS officially in Hyper-V.

I am looking for a provider that will provide an equivalent of DID/DOD trunks via IAX, IAX2 or SIP using numbers in Metro Chicago (prefer Skokie), LA (prefer West Hollywood or Venice), and/or Orlando (prefer Winter Garden). If I can migrate some of my existing numbers using LNP, that would be even better, but it is not a requirement. While I know that there are several companies that will

XCP Users: I received an email from Alberto Gonzalez Rodriguez who created OpenXenCenter/OpenXenManager and is now working on OpenXenWebManager. Info to access the tool is: steps to test are: download from http://bit.ly/bres2U uncompress: tar xvfz xenwebmanager_rev24_full.tar.gz cd xenwebmanager python frontend.py and with a browser open http://localhost:8080 (or http://ip:8080) you need

Hi, Kerry Garrison from Fonality will be with us live to address the Trixbox so-called "phone home" script issue. I'm going to try to have something about the year "2007 in review" for any and all VOIP and Asterisk-related events, so if anyone wants to report on what they've been doing in 2007, you're welcome to chime in. We're also talking about doing a

Hi, This week we are pleased to welcome Andy Abramson (http://andyabramson.blogs.com/voipwatch/) as our guest. Andy is one of the most avid observers of the world of VoIP, from Asterisk and its variations to all kinds of ramifications of VoIP. I'm sure we'll pass a lot of the VoIP News in review tomorrow at the usual time and place: Friday 18 Sept at 12 Noon EDT, your local time:

Hi, We're pleased have a 25-year telephony veteran with us tomorrow, Aswath Rao. Aswath maintains that "Trapezoidal VoIP is Evil". Join us and ask questions, make comments, argue about geeky details... and maybe win a Gigaset S675IP SIP/DECT g722-capable phone with an additional handset. Those of us who have these phones like them a lot. All dial in info is here: http://VUC.me -

Team: It is time to start thinking about our 2nd XCP meeting for the Xen.org XCP team. The last meeting was held on May 25, 2010 at 14:00 UTC. Notes are at http://wiki.xensource.com/xenwiki/XCP_Monthly_Meetings. For the next meeting I hope to schedule a time that is better for our Asian participants so please contact me with a time that would be best if you live in Japan, Korea, Australia, etc.

Team: It is time to start thinking about our 2nd XCP meeting for the Xen.org XCP team. The last meeting was held on May 25, 2010 at 14:00 UTC. Notes are at http://wiki.xensource.com/xenwiki/XCP_Monthly_Meetings. For the next meeting I hope to schedule a time that is better for our Asian participants so please contact me with a time that would be best if you live in Japan, Korea, Australia, etc.

Tomorrow, Friday August 21st at noon EDT, we'll be on the rampage again with the VoIP Users Conference, much of which concerns Asterisk and the Asterisk community. We've covered a lot of ground in the past 2 1/2 years, and had a lot of great guest presenters. In the meantime, there's a new gadget to test at PhoneFromHere.com, you can hear the wideband demo on the ZipDX.com bridge:

As usual, the BerkeleyTIP group programming project is to learn about & work toward implementing our own Asterisk VOIP conference server, during our meeting. 10A-6P Pacific USA (-8H GMT) = 1P-9P Eastern US = 6P-2AM GMT. http://sites.google.com/site/berkeleytip/ Join the VOIP online conference & help out, or chat with your buddies. :)

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