similar to: [Asterisk-Dev] Cluecon, A mix of leading Open Source VoIP devlopers...

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

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

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

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

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

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

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,

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

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

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All, is anyone using Bayonne in conjunction with Asterisk? I'm currently using only Bayonne, but I'm investigating the possibilities of switching the telephony frontend over to Asterisk, and have Asterisk route the IVR tasks to Bayonne through H323. Anyone care to share his views on this approach? Any pointers or do's and don'ts? All info is greatly appreciated! Regards,

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

Hi all, I want to deploy a SIP Proxy but I just don't know which one to choose. Researching in the Internet I found the following ones: * SIP Express Router <http://www.voip-info.org/wiki/view/SIP+Express+Router>: SER is used by many SIP providers standalone or in conjunction with Asterisk * Vovida.org <http://www.voip-info.org/wiki/view/Vovida.org> * sipX

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, Anyone know anything about asterlink? I've heard good things about them. So I thought I'd give them a try. I went to sign-up.... however, I haven't heard from them, I've called them several times between 9am and 5pm EST and no responce. Called today again after 9am EST, no responce. Tried Sales, Billing, and CS, and got the same person's voicemail! Is this a oneman

I found an old Dialogic card in an abandoned PC, that I think is a Dialogic D/41EPCI based on some googling.. The lspci output says: 00:09.0 Bridge: PLX Technology, Inc. PCI <-> IOBus Bridge (rev 01) Subsystem: Dialogic Corp: Unknown device 0529 I'm just getting started with Asterisk to build a SOHO system. Any suggestions for what I can do with this card? It would be

Everyone we have a few new things to give back to the asterisk community. http://bugs.digium.com/bug_view_page.php?bug_id=0002379 http://bugs.digium.com/bug_view_page.php?bug_id=0002380 http://bugs.digium.com/bug_view_page.php?bug_id=0002381 These include app_chanspy, the ability to spy on ANY bridged call taking place inside asterisk. NOT just ZAP as with ZapScan/Barge. Native format_* files

We have three remote call center Asterisk servers communicating with two central Asterisk boxes over a private IP-VPN with QoS. All systems were running Asterisk 1.0.7 communicating via IAX2 with little or no quality issues at all. Once we upgraded to Asterisk 1.2.4 call quality with IAX2 was horrific. We tried with/without jitterbuffer. We messed with every jitterbuffer parameter. We tried

Hello, I am very new to Asterisk, so please forgive me for my basic question. I couldn't find a good answer to this on the info boards. I'd like to know if the following is possible and reasonable to accomplish: I'd like to configure a voice recording system using Asterisk and a Tormenta2 Quad T1 card. A co-worker was able to create this system a while back with Bayonne and a