similar to: Friday June 5th @12 Noon EDT: Sipgate invades the USA, more HD Voice, Video chat

Hi all, In about 4 hours from this writing, the G.722 conference bridge will be brought up, the Talkshoe G.711 also, so you can call in via SIP, PSTN or Skype (experimental) http://vuc.me for all the gritty details IRC #voip-users-conference anytime today We'll also be talking about hosted PBX systems and what they might need to bring telephone into not the 20th, but the 21st century. It

Today, Chris Matthieu, Founder & CEO of GetVocal, entered the cloud-based communications market in February, with its launch of Teleku. Teleku is a new cloud-based telecom service that allows Web developers to build and host phone applications that answer inbound calls and initiate outbound calls, interact with Web applications, and send/receive SMS text messages! Chris, who you may have

Yes, we mostly talk about asterisk, hardware, software, phones, people, events and things asterisk-related. Asterisk is a registered trademark of Digium. Asterix is a registered trademark of Ren? Goscinny. We never discuss that, though. Here's the short URL for sending out to others that might be interested: http://x2z.eu It lands here: http://www.VoipUsersConference.org Once again, I

Hi, This week's VoIP Users Conference will be tie the Talkshoe PSTN/SIP ULAW conference bridge to the ZipDX.com G.722-capable bridge. It may be a little crazy, but I'm looking forward to getting some explanations from David Frankel about the effects of wide band (or as Polycom calls it, "HD Voice") on conferencing. David can probably answer any technical questions you may have

Morning! This may be the "day of the dead" in some regions, but we expect the usual lively discussion today at 9AM PDT, 11 Central, 12 Noon EDT, 4PM UK and Portugal, 5PM Paris, $deity-forsaken hour down under. This Sunday, I believe the USA falls back to Standard time. Future VUC are still at 12 Noon EST. Info site: http://voipusersconference.org PSTN (724) 444-7444 enter 22622# 1#

Hi all, You may have heard yesterday that Adhearsion and Voxeo have created a new baby, Voxeo Labs. From our (non-biz) point of view, I'd recommend following the blogs: http://blogs.voxeo.com/ to see how what they do might be of interest to you and your asterisk/voip activities, commercial or private. Since I myself know little about what this all means, I've invited a lot of bright

Hi all, You're welcome as always to join the talk on the VoIP Users Conference, VUC for short. VUC began as the "Asterisk Users Conference" but for obvious reasons, we changed the name in the first year, although Digium was our sponsor for three years. We still have plenty of you who are asterisk users and developers on this our fifth year. Come on by, listen, talk ro text on IRC.

Hi, As usual, you can get all the dial in information at http://VoipUsersConference.org IRC is on Freenode.net #voip-users-conference join this even if you can't call in. Call via SIP: talkshoe at vuc.onsip.com (thanks to OnSip.com) Call via PSTN (724) 444-7444 DTMF 22622# 1# or try this: 7463#22622#1 at proxy.ideasip.com (thanks to IdeaSIP.com) or to just look up talkshoe server IP:

Asterisk Users Conference Fri 12:30 PM EDT ** see http://x2z.eu for instructions on how to join or listen RussellB from Digium and others will be around to tell us about the recent devcon and hopefully answer any questions about the near future of asterisk branches, what's new, etc. Hope you can make it! ** exten => AUC,1,Dial(SIP/123@66.212.134.192,60,D(22622#${YOUR_PIN}#))

Hi, Quick reminder that the conference is happening today at 12:30 PM EDT. I'd like to talk more about updating to 1.4. I now have a test box running asterisk 1.4.5, CentOS 5 and Lumenvox speech rec software. Seems to be fine except for some double NAT issues that could be router specific. Byran Johns from Shelton-Johns is our guest to share some of his extensive experience. More about him

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:

Hi, I met Michael Iedema of the Askozia pbx project at ANOOMA (ANOOMA is the Cannes Festival of the Asterisk world) and begged him to take time off from porting Askozia from FreeBSD to linux to tell us about his project and the problems of porting as well. Hope you'll all be able to stop by and lmisten or download or stream the recordings. We are having a lot of people call in on widenabd

Hi, Like me, some of you probably remember Jim as one of the pioneers along with Leif and Jarod. These guys "wrote the book", literally. Jim is our guest tomorrow and he'll be talking about system building, among other things. We always have a good time AND get stuff done on the Conference so come by and join us: Silently, on IRC: #voip-users-conference (freenode.net) - stop by

Hey, This week I return from AstriEurop and I'm lucky because I left yesterday, many of my Asterisk friends are stranded because of the air and rail strikes that got worse today. I will mention a few things I think are interesting from my meetings there and hope a few others will make it and share their experiences. in particular, Mark Spencer's keynote held a surprise or two. Also, we

Hi, If I was going to post this as an iPhone-only SIP client, I'd expect loud booing and hissing, but Media5 mobile SIP client is available for the Symbian S60 platform, too, or will be shortly. Interested? To join us and hear about Media5 form Pascal Dore, see http://vuc.me Speaking of mobile, if any of you know the people who are developing the SIP client for Android, we'd love to

By popular request, we've convinced someone from the VoIP Abuse Project to join us tomorrow at noon on VUC. I think many of you will be interested in this topic, so please come by, join in and ask questions. http://vuc.me for all connection info and links to VoIP Abuse Project A couple of other features and announcements are scheduled after that segment, including an Astricon update and

Friday we'll be hearing about SIP Communicator Java VoIP and Instant Messaging client. SIP Communicator is an audio/video Internet phone and instant messenger that supports some of the most popular VoIP and instant messaging protocols such as SIP, Jabber, AIM/ICQ, MSN, Yahoo! Messenger, Bonjour, IRC and a whole lot of other useful features. Open Source / Free Software, and is freely available

Hi All, Today on the VUC, we'll be welcoming Sipgate CEO Thilo Salmon back to tell about their choice of partners in their latest services. I will be announcing the first #VUC VoIP & Tell discount code for Astricon in Denver, October 25-27. Join us on sip:200909 at login.zipdx.com using g722 or g711 codec or see http://vuc.me for the other ways to connect. Hear you there! /r

Nir Simionovich is about to become a father. He will be joining our conference at 12 Noon EDT today from the Maternity Ward to talk about Amazon EC2 cloud computing with Asterisk. Nir gave a very good presentation on this at AMOOCON a few weeks ago (see http://www.amoocon.de for more on that). The advantage here though is that he'll be live with us for your questions. All the details on how

Hi, Tomorrow's guest is Mike Seto from Polycom who will be "grilled" on application development for their phones. Coincidentally, there is a contest for the best microbrowser application. Anyone can join in on this: http://tr.im/WinPolycom Join us in the usual places: IRC #voip-users-conference Call Info: http://www.VoipUsersConference.org SIP URI: 7463#22622#PIN at