similar to: AstriCon videos: a question of method (Robin)

Hey guys, Having a little gpg issue I was wondering if someone could help me with. A friend of mine sent me an encrypted message. So I searched online and found a a set of keys that correspond with his email address. And imported them. But when I go to decrypt the message, this is what I get: [root at ops:~] #gpg --decrypt roger-message gpg: encrypted with 2048-bit RSA key, ID 9617EA5C,

I'm doing some quick research on how to get our videos from AstriCon available in a "reasonable" format that allows easy viewing, reduces our bandwidth costs, and allows good tracking for who/where/what is viewing the videos. YouTube seems to have a very nice set of tools and statistics collection methods, and might be perfect EXCEPT.... Their main limitation right now

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

Ok I have to get a vote of all the people that are going to come to Cluecon so we order the beer keg's for the developers board room. Anyone have any preference? (if you haven't registered for ClueCon now is the time to register!) Choices... choices... choices... I want Red Bull on tap! /b --- Anakin: ?You?re either with me, or you?re my enemy.? Obi-Wan: ?Only a Sith could be an

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

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

Hi all, I'm hoping that someone has seen this one before. I am trying to get Civ4 working in Wine 1.2 on my Mac and I get so far as the stage where it initializes Python and it fails. I've attached the log at the end of this transmission. It's relatively short as these things go. I can see at least one error: 'import site' failed; use -v for traceback but I have no clue as to

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

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

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 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

I want to interface asterisk with a legacy pbx that has around 23 extensions through my 8 fxs card, how do i work around this? Hint: I have already terminated 8 extensions from the legacy PBX, i was thinking whether i can peer the extensions from the PBX i.e like 5 extensions be peered to one extension connecting to the fxs? How can i do this? Thanks in advance, -------------- next part

August 5th, 2005: Craig Southeren announced today that OPAL (http://www.voxgratia.org) now provides support for the IAX2 protocol(Written by Derek Smithies and released under the MPL). This support allows you to use chan_woomera (http://www.pbxfreeware.org) driver developed by Anthony Minessale II to interconnect your asterisk systems and use the IAX2, SIP, and H.323 protocols. I would

Hello everyone, I have just finished up work on a new release of AstLinux, the embedded/live cd/minimal built from scratch Linux distro centered around Asterisk. Most of the work for 0.2.8 has been on the ISO image (live/install cd), the init system and further system customization with new rc.conf variables. - The ISO image now allows you to install AstLinux to a local disk/CF card,