similar to: Friday Jan 30th at 12 Noon EST: Design the Phone of the Future

Hi, Few subjects cause as many arguments as "which SIP client works best?" on IRC #asterisk, voip forums, and probably the -users mailing list. I have tried most of the SIP clients available in the last 5 years, both with Asterisk and other platforms such as OnSIP.com, IConnectHere.com, ZipDX.com and the venerable old FWD (in the days when that almost worked). Speaking of ZipDX, we

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

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

Randal Schwartz (FLOSS Weekly) joins us for this slightly OT discussion about Open Source software and its role in all of our lives. You're all welcome to join in as usual. We will also be connected to a g722 bridge for more experimentation with the merits of wide band audio. This should give you all the info needed : http://www.voipusersconference.org and jump on IRC.freenode.net on

This Friday, February 15th, at 12 Noon EST, 9AM PST, 17:00 UTC, Lumenvox will be joining us on the VoIP Users Conference. This week, the last in a series about IVR, Lumenvox will be there to discuss and field your questions on their speech recognition solutions. http://www.VoipUsersConference.org - for info on the conference, how to connect, etc IRC freenode.net #voip-users-conference - to

Hi, Our guest this Friday is Himanshu Dwivendi, author of the book Hacking VoIP. You're welcome to come discuss it with us on the conference. Find your local time by going to http://vuc.me/next - the conference begins a little before 12 Noon Eastern Time. VUC has an IRC channel #vuc on Freenode.net and uses Skype for Asterisk to allow connections to listen and/or speak. SIP g722 : 200901 at

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

Hi all, What is Bufferbloat? http://gettys.wordpress.com/bufferbloat-faq/ Maybe this kind of discussion will bring out the John Todds of this world, I can only hope and dream: Bufferbloat: http://www.voipusersconference.org/2011/bufferbloat/ Call in and talk to Jim Gettys, who co-developed X Window System and was a part of HTTP/1.1 - this is someone we'll all be proud to meet, and you can

This Friday's edition of the weekly VoIP Users Conference call is all about wideband audio (aka HD Voice) and conferencing. The guest for this call is David Frankel, CEO of ZipDX a commercial service that specializes in wideband conferencing. We expect an interesting call touching on many aspects of VoIP going beyond the traditional phone service, conference bridges, technical standards,

Hello, This Friday March 4th is the 307th VoIP Users Conference. In a few weeks, we'll be starting our 5th year of this weekly live event that began life as the Asterisk Users Conference. We'd love to have you join on on our call with this week's guest OnSIP.com by connecting via SIP:200901 at login.zipdx.com Skype:vuc.me PSTN: +1 567 252 2286 iNum: +883 5100 123 94882 Tex backchat

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.

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, 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 -

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

Hi all, We've had Yusuf Motiwala from TringMe on the VoIP Users Conference before when he annouced their Flash-based web phones. Now they've come up with something that tantalizes me, VoicePHP. Sure XML is a standard and fairly easy to implement, but not as easy as PHP, which I have used since version 2 (Yup, shadesof.phtml) Here's GigOm's take on it: http://tr.im/2y4g Check

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:

Hi, December 5th, 1999 was the initial release of Asterisk by Mark Spencer. We'll be celebrating this by gathering as usual at 12 Noon Eastern (9AM Pacific, 10 MST, 11 Central, 5PM UK and Western EU) for the VoIP Users Conference. You can get all the dial in information at http://VoipUsersConference.org including info on a SipAddHeader() kludge to avoid DTMF problems. IRC is Freenode.net

Every week we try to get guests with ideas, products and services you haven't had time to check out to come and talk about what they're doing. Tomorrow, Pika Technologies will be with us. Friday, March 7that 12:00 PM (Eastern US) 9AM PST, 5PM GMT *** Call (724) 444-7444 or SIP:123 at 66.212.134.192 *** After the call connects, enter the conf: 22622# and your_PIN# (or 1# if you

TODAY, Friday January 4th at 12 Noon EST, 11 AM Central, 9AM Pacific, Mountain figure it out, 17:00 UTC Mark joins us to talk about IAX, the appliance, what's new in the asterisk worldwide communities and answer any questions you may have. Why not take this opportunity to ask questions or make comments? This conference is the largest *live* online meeting of asterisk users in the world. Each

Hi, Lots of VoIP, SIP and Asterisk-related discussion and some free phones and Polycom software today, join us at the usual place: http://www.voipusersconference.org Call sip:200901 at login.zipdx.com Skype:vuc.me (via Skype for Asterisk and PhonefromHere.com) Listen Live 16khz mp3 Stream: http://vuc.li/mp3stream None of the above will work until about 15 minutes before. The mp3 stream goes