similar to: VUC Today at 1 PM EST: Counterpath/Bria

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, 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

Hi, We had a last-minute cancellation from Vivox for today's conference. It happens that someone suggested a guest idea, Howler Technologies CTO Jay Fenton, who agreed to join the call from the road. Anything you want to know about transcoding to and from g729 is out topic for the first hour. My pal David Duffet knows this technology well and has kindly signed in to help guide us through this

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

Interesting offering, free from Twilio, this is php you install on your own server to build a brandable "VBX". Worth checking out! Listen to tomorrow for more about this and talk to lead engineer or Twilio CEO if you have any questions; sip:200901 at login.zipdx.com or Skype:vuc.me IRC: #vuc on Freenode.net or http://vuc.me/irc Info about VUC is htp://vuc.me Best, /r

I have a pair of Asterisk 11.5.1 servers operating as a load balanced cluster, with DNS SRV records set up to weight them 60/40 relative to each other (both at priority 0). The back-end is MySQL Realtime, and everything works pretty well with the Cisco SPA phones & ATAs that represent the majority of my endpoints. I recently tried to add an iPhone with the Bria softphone application, to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Everybody, I am experiencing some troubles with my Bria iPhone Edition (v. 1.2.8 build 5312, on iOS 4.2.1 iPhone 3G) and Asterisk 1.8.3.2 + TLS/SRTP on LAN (without NAT). With 2 computer clients (Blink, one on Mac, one on Windows/Linux),9i can have a very fine secure conversation in both directions. When I want to do the same with my iPhone,

This should be interesting, a double header Friday at 12 Noon EDT, session 2 at 1PM EDT. 1) Pascal Dor?, Media5corp. Pascal will talk about what they've been up to in the year since his last visit. Thanks to the Asterisk mailing list and VoIP community, their Media5fone was able to fix its g722 implementation. I like their product a lot and used it extensively on my old iPod Touch to make and

Today, sessions 320-321 of the VoIP Users Conference will take place at the usual time, 12 Noon Eastern [ http://vuc.me/next for local times ] We'll be talking to Sangoma's Frederic Dickey about NetBorder 4.0. You can download or watch his accompanying slide presentation here: http://vuc.li/Sangoma-2011 Next, Pascal Dor?, mVoIP chief over at Media5 Corporation will talk about the

Hi, I'm excited to announce that the guys from AG Projects are stopping by for a beer tomorrow on VoIP Users Conference, aka VUC. You should already be familiar with their excellent multi-platform SIP client, Blink (http://icanblink.com) While Adrian and Sa?l enjoy a few exotic brews with us, they'll also be telling us about SylkServer: - creation and delivery of rich multimedia

Hi, Today at 12 Non EDT, Dan York will be with us to talk about the recent on and off moments of Google Voice SIP URI calling. Like Skype + Asterisk (or any SIP), Google Voice and SIP compose the other shoe waiting to drop. We're following this with interest. So GV turned on SIP URI and then a few days later, turned it off. Why? Did the geeks (like us) jump on this too quickly or too heavily?

Hi all, Friday at 12 Noon EDT, we'll be talking to Emil Ivov of Jitsi.org (formerly SIP Communicator) and Thiago Rocha Camargo (of Nimbuzz) about Jabber, something the Asterisk community is becoming more interested in by the day. Join us to learn more about Jabber and SIP or to share your knowledge and experience. As always, the VUC discussion includes people from very diverse backgrounds, so

Our guest today has a long and interesting background in network and VoIP technologies as well as having been at the head of Digium's product management: Bill "@beelinebill" Miller. To join us and hear (and talk to) Bill, see http://vuc.me You can call sip:200901 at login.zipdx.com in g722 wideband (or g711) Jump on IRC #vuc channel There are a number of other ways to connect

Hi, Tomorrow, our discussion is around iNum with lots of interesting people chiming in, including the Voxbone people who manage the space. If you ever wondered about iNum and why you might care about it, how it works, who offers it and who actually uses it, here's a chance to find out more. Join us live at 12 Noon EST or find the local time for you at http://vuc.me/next Call: sip:200901 at

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

Topic: Phono and Phono SDK, an exciting development you can read about at http://phono.com You can learn all about the technical aspects of Phono and the SDK Friday with Chris Matthieu, but here are a couple of interesting implementations that don't require much effort to show a proof-of-concept: 1) There is already a WordPress plugin that literally allows you to add a button on your

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

Quick reminder before Astricon (from which we will be reporting from live): Tomorrow's guest will be VoIP author Alex Robar. Alex has worked with open source telephony solutions for the past four years, and has collaborated on the development and growth of an international Asterisk-based VoIP peering network. His book is FreePBX 2.5 Powerful Telephony Solutions and we'll be chatting with

Hello, In about one hour we should be chatting with Tim Behrins of Voxbone about their initiative, iNum. I say "should" because he's the scheduled guest, but I haven't heard from him today :) Next week, we'll be "Hacking VoIP" Feel free to top post your answers, it seems to stimulate conversation. /r http://VoipUsersConference.org for the usual data or jump on

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.