similar to: Friday Dec 19th at Noon ET: Jazinga pbx appliance

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, 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

I don't understand why so many government sites fail to provide some sort of feed to their daily bulletins. What I am venting about in specific are the Canadian CRTC and FCC sites, every day I have to go to the website and when I reach the content, usually it isn't even HTML but a Word or PDF file. So finally today I decided to do something about it and wrote a little app that scrapes

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 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, 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

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 List, Does anyone have experiences to relate on the various Asterisk-based PBX appliances out there? Like the Aastra 160, Digium S844i, etc. Do the Epygi Quadro and Grandstream GXE also use Asterisk? Thanks, Femi -------------- next part -------------- An HTML attachment was scrubbed... URL:

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#

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, 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

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)

The subject of tomorrow's VoIP Users Conference will be mobile VoIP. If you have any interest, please join us. I myself am tesing a bunch of iPod applications to use with all the usual suspects: OnSIP, Sipgate, Gizmo, Skype, your asterisk box, etc. Details for joining the call are are at http://VUC.me or http://VoipUsersConference.org IRC: #voip-users-conference on Freenode.net See you

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

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

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

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

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"

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

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