similar to: Friday at 12 Noon EDT: VICIDIAL

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

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

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 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

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 -

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

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 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, This week we are pleased to welcome Andy Abramson (http://andyabramson.blogs.com/voipwatch/) as our guest. Andy is one of the most avid observers of the world of VoIP, from Asterisk and its variations to all kinds of ramifications of VoIP. I'm sure we'll pass a lot of the VoIP News in review tomorrow at the usual time and place: Friday 18 Sept at 12 Noon EDT, your local time:

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, 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, 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

Please join us today at 9AM PDT, 12 Noon EDT for the VoIP Users Conference to talk about the latest news and events in the wonderful world of VoIP. IRC #voip-users-conference SIP 7463#22622#1 at proxy.ideasip.com for g711 SIP 200901 at login.zipdx.com (for g722 wideband-capable devices) See http://VUC.me for more details /r

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

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

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

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

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