similar to: Asterisk and EC2 today at 12 Noon EDT

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

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

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, 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

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, 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

Already Friday, the week went by in a Flash. If you haven't yet registered for a free Sipgate DID, I suggest you go do so. If you are in, or interested in the business, Sipgate has a few tricks up their sleeves and you should be aware of them. Someone from Sipgate will be joining the conference, which begins at 12 Noon EDT (9AM PDT, 10 Mountain, 11 Central, 5PM UK, 6 Central Europe) ZipDX

Hi, I met Matt Florell at AMOOCON and tried to record an interview. I was pleased with the results, but later found that the battery deleted the audio file when it went dead. Today, we'll have Matt live to talk about VICIDIAL and answer any questions you may have about it. For more on this: http://VUC.me IRC #voip-users-conference on Freenode.net Call 7463#22622#1 at proxy.ideasip.com to

http://vuc.me Kamailio, Open SER and Asterisk walk into a bar... The bartender is Alex Balashov, someone whose posts I have long admired on this list. Alex has agreed to take us through the following areas: - Relationship of Kamailio to OpenSER project history. - What is Kamailio/OpenSER? - SIP proxy - SIP server (for certain purposes, such as registrar, presence user agent, etc.) -

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 -

To celebrate three years of the VoIP Users Conference, we're doing a 24-hour VoIP conference call today. Details are at http://voipathon.org IRC: #vuc on Freenode.net SIP: voipathon at vuc.onsip.com - Enter 22622# and your PIN# if you have no PIN you can listen using 1# iNum - +883 51007 039 9924 PSTN: +1 724 444 7444 again, 22622#1# or PIN# if you have one. Those of you in the

Hi, I hope everyone will have a joyous holiday time. If you happen to find yourself at loose ends on Friday after the eat and drink fest that many of us will be privileged to enjoy Thursday, come and talk with us about Skype for Asterisk. Steve Sokol, who's in charge of the beta for this module at Digium, has courageously agreed to give us a report on how it's coming along. Spread this

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, 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 all, Get your questions ready as tomorrow's VUC call will feature Shidan Gouran, CTO of Jazinga, makers of a new Asterisk appliance. Jazinga have developed a web 2.0 GUI for their embedded Asterisk appliance. We all love GUIs, right? They want to make it easy for a non-techie to setup a small office Asterisk solution. For details about the Jazinga product you can see Michael Graves'

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, Plenty to talk about today. Dave Nelsen, a founder of Talkshoe, has a lot of experience in the telecommunications space and he joins us today to chat about its current state, conferencing and whatever else comes to mind. So we have a meta conference aout conferencing, it won't be the first time :) You probably saw John Todd's message on one of the lists: Skype for Asterisk is in open

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 All, At the usual time, 12 Noon ET on Friday April 3rd, we expect Michael Robertson to join the discussion to filed questions about OpenSky and Gizmo5. I have been testing all of these Skype to X methods except SIP for Skype since I have no word from them. I can tell you that we've had good results with bith Skype for Asterisk and OpenSky. In fact, I am currently accepting calls to my