similar to: ClueCon, Vote?

ClueCon is coming in 2 weeks so we urge everyone who plans on attending to register today so we get a proper headcount! ClueCon was put together by Asterlink, the same team of people who helped shape Asterisk into what it is today by writing features, fixing bugs, offering IRC support and assisting with the management of the development effort. We have produced several real-world solutions based

I'm relatively new to Asterisk and I'm hoping attending Cluecon will be a good way to get up to speed on the project and hear about what others are doing with it. We currently use a Cisco IP phone system at my office, although I just added an asterisk box to provide soft phones to our travelling users. (IAX2 is a lot easier to get through firewalls than cisco's protocols). Terry

Peter Nixon will be making the trip to Chicago to speak at Cluecon, he'll be speaking on the topic of "Real world deployment of Open Source". Peter has done tremendous amounts of work on the FreeRadius project. In addition if you're wanting to get sponsorship in this is the week to do so, we are sending everything off to the printer to get printed by friday. Thanks,

Digium, the creator and primary developer of Asterisk, the industrys first Open Source PBX, will be hosting a pizza party from 4pm to 6pm on the first day of Cluecon. We look forward to everyone coming out to enjoy this opportunity to meet fellow developers and users in a more casual environment. I would like to personally thank Mark Spencer and Digium for their support. Thanks, Brian

Robin, Thanks for the viddler.com suggestion! I'm uploading all of the ClueCon videos to it right now. John, so far I'd have to give viddler.com two thumbs up. I'm adding my stuff here: http://www.viddler.com/explore/cluecon Your ClueCon presentation should show up some time on Friday. I've noticed that there's a little bit of a lag time between upload and video being

Through the generosity of our Premier sponsor, Sangoma Technologies we are proud to welcome Craig Southeren all the way from Australia. Mr. Southeren.s work has pioneered the development of open source telephony applications with his ground-breaking OpenH323 protocol stack that stood alone as the only open source VOIP software for quite some time. Today, Craig continues to raise the bar

Just an FYI guys we have some of the leading open source developers and projects going to speak/showcase at Cluecon. These include: Mark Spencer - Asterisk Bob Andreasen - SIPFoundry Craig Southeren - OpenH323 David Sugar - Bayonne This should be an exciting event for all. Register Today! hope to see you there! Brian West Asterlink.com _______________________________________________

Vikrant Mathur is the lead developer for the open source OSP Toolkit available on SIPfoundry. Mr. Mathur began his career in telecommunications as a software engineer at Hughes Software Systems where he focused on softswitch development. After completing his Masters degree in Electrical Engineering at North Carolina State University he joined TransNexus as a senior software engineer

Vikrant Mathur is the lead developer for the open source OSP Toolkit available on SIPfoundry. Mr. Mathur began his career in telecommunications as a software engineer at Hughes Software Systems where he focused on softswitch development. After completing his Masters degree in Electrical Engineering at North Carolina State University he joined TransNexus as a senior software engineer

I would like to send out a reminder we do still have a few rooms available for Cluecon Aug 3rd thru the 5th if you're still wanting to register and go please do so ASAP. Thanks, Brian

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Question: is anyone planning on going to the Cluecon convention this year? (www.cluecon.com <http://www.cluecon.com/> ) I'm hoping to go this year and I'm hoping to meet other OSS telephony users and developers. BTW, Anthony Minessale said that there is a need for Asterisk speakers, so if you're an Asterisk user (or expert) and you're in the Chicago area in early August

Regarding my earlier post about labels and the 'n' priority: The TFOT book covers the use of these. See the box on page 81 entitled "Unnumbered Priorities." http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip -MC

Hi, We are currently developing an LLVM target for a micro-controller, and would like our TargetLowering::LowerCall method to emit PUSH instructions (instead of STORE) to pass arguments (which would improve code density for function calls). Is there a way of keeping track of the stack pointer changes implied by the PUSH instruction to calculate the correct offsets in

I am using Ubuntu Hardy, using the latest release of wine from the wine repos. I have an nVidia 7600GT with 256MB of video memory. One windows game i installed, Medal of Honor Allied Assault, has a simple config utility that sets game settings based on your hardware. This utility is detecting only 64MB of video memory. I tried setting it using wine regedit, HKCU>Software>Wine>Direct3d

Anyone have any insight as to why voip-info.org has been up and down all day, and more importantly unreliable for the last month? I assume the bandwidth is being donated or something, but surely someone would be willing to donate reliable bandwidth as the knowledge hosted on the site (which is also donated!) is worth way more than the bandwidth. There is no doubt it is the best

Hey guys Ok i was curious to see if Medal of honor would work 2010 tier 1 edition would work I installed the dvd just fine. a little buggy on the blue bar but hey xD it installed fine At first i couldnt run the game Then i figured out that you go to your home folder and find the MoH folder. edit the settings and simply changed "directx mode" to 9 not auto And it ran, starts up fine

Hi, I was wondering if we can check the voicemails remotely from a cell or a landline number. We have SIP 3 Digit Extensions connected to Asterisk server. If users are away from Desk & need to access voicemails can they dial in to Asterisk PBX & check their messages. I know one can check through web link & even have mailed. Aslo I have checked regarding DISA, but I am not kind of

Hey gang, I've been able to use sipp to produce some call volume on our asterisk server. The server has no problems handling 50 simul calls. But then again, no RTP is being done. I tried to use the rtp echo ability of sipp but that doesn't seem to work right. I also setup a fake number in asterisk that when called by sipp, would dial another number via PRI, hoping that some 729

Hi, Thanks a lot for your answer. It seems lld is still the best solution, even if it does not work "right out of the box" for us today. We already have a solution for the "objcopy" part (added the required output format to llvm-objdump). The ScriptLayout class seems to be empty for now (on the master branch at least), but we do not need linker scripts today. All that is