similar to: Friday Feb 6th at 12 Noon EST: Polycom and Application Development

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

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, 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, We're moving ahead on this, there are some interesting files to check out thanks to davevg's perl skills. If you find a way to write an interesting application to run on a Polycom, you can win a Polycom IP 450. Even if you don't care about that, you may be interested in knowing more about programming for the Polycom. These phones are way underused IMO. Join us for a discussion of

Happy coming Spring, 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. Friday, February 22 at 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 have no PIN)

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

Our guest is tomorrow Mobeen Khan is Chief Operating Officer of Metaphor Solutions who offer "Plug & Play IVR On-Demand" http://www.metaphorivr.com Instructions to join the conference: http://VoipUsersConference.org IRC: freenode.net #voip-users-conference The weekly Friday Noon "VoIP Users Conference" call was originally the "Asterisk Users Conference" but

Hi, Quoting Digium's blog post (blogs.digium.com): "The market talked, we listened, and today we?re ready to take calls, answer questions and do whatever needs to be done to make Asterisk work for our customers. It is now safe to step into the Open." John Todd and Steve Sokol will be online live at 12 Noon EDT (9AM Pacific, 10 Mountain, 11 Central, 5PM UK, and 6PM in Western

Hi, When Jason Goecke talks, VoIP ideas become reality, and this makes my day. On this call we?ll talk about the newest features in Tropo and how to get started with telephony apps in the cloud without adding new infrastructure. Here's a chance to speak directly to Jason (or JSON as we now call him) to ask your questions about VoIP cloud apps. Tropo is a cloud communications platform that to

Hi, Allison Smith continues to contribute to the open source asterisk resources and she is launching a new site that will make it even easier to grab sound files. Allison joins us to talk about that and whatever else comes up. She says: "I have been the Voice of Asterisk -- the world's fastest-growing telephony platform -- since its inception (which for me, was marked by an animated

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

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

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.) -

Fridays @ 12 Noon EST - 9 AM PST - 17:00 GMT etc ad infinitum THIS WEEK Among other topics, we'll be hearing about TringMe (http://www.tringme.com) a Phone Me solution that works for just about anyone with audio and a headset. Anything else about VOIP is permitted, but the following words are permanently banned from the conference: "Intermittent", "Unable to reproduce"

This Friday as usual at 12 Noon EST, 9 AM PST, 17:00 UTC Staying in the IVR domain for one more week, our guest today is Allison Smith who literally needs no introduction. I have a couple of secret tapes of Allison that she knows nothing about from the earliest days of her career. Instructions to join the conference: http://www.VoipUsersConference.org IRC: freenode.net #voip-users-conference

Some of us have specific ideas about this (hi dave), whatever yours are they'd be appreciated in the conference. Also, ZipDX is again letting us use their g722 bridge connected to our Talkshoe bridge. Was hoping for a Skype bridge from Steve Sokol's beta, but no news from him ATM. IRC: #voip-users-conference How to participate: http://www.VoipUsersConference.org Twitter: voipusers

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