July 3rd: Twilio, Talk to the Cloud
Posted by admin in Future Topics, VoIP on July 1st, 2009
Twilio has an application that will call you on July 3rd and remind you to join us a few minutes before this call. Fill out this form to try it and get a special greeting!
Thanks to Dave Michels, who met him at Glue Con, Twilio CEO Jeff Lawson and CTO Evan Cooke join us to discuss their “cloud telephony” offering. You can prepare by getting a free account and checking out what can be done with the platform. Note that speech to text (transcription) is a part of it.
“Twilio provides an in-cloud API for voice communications that leverages existing web development skills, resources and infrastructure.”
Matt Florell VICIDIAL
Audio clip: Adobe Flash Player (version 9 or above) is required to play this audio clip. Download the latest version here. You also need to have JavaScript enabled in your browser.
Spend an hour and a half with Matt Florell talking about VICIDIAL, the Open Source Call Center software, and plenty of other subjects including CentOS, Ubuntu, resource issues, hardware for robust asterisk systems.
About VICIDIAL:
VICIDIAL is an enterprise class, open source, call center suite in use by many large call centers around the world.
VICIDIAL has a full featured predictive dialer. It can also function as an ACD for inbound calls, or closer calls coming from VICIDIAL outbound fronters. It is capable of inbound, outbound, and blended call handling. VICIDIAL even allows you to have agents logged in from remote locations.
June 19th: Nir Simionovich on EC2
There is a full text transcript of this hour below.
Part 1: (Talkshoe)
Audio clip: Adobe Flash Player (version 9 or above) is required to play this audio clip. Download the latest version here. You also need to have JavaScript enabled in your browser.
Part 2 (ZipDX recording served from CloudFront):
Audio clip: Adobe Flash Player (version 9 or above) is required to play this audio clip. Download the latest version here. You also need to have JavaScript enabled in your browser.
We’ve tried to get people to talk about this before, but we didn’t get a deep explanation. Here’s a chance to ask Nir, who did a compelling presentation at AMOOCON on Asterisk on EC2, to clear up any of those nagging questions you might have.
Nir’s AMOOCON presentation, Dynamic Asterisk Scalability with Amazon EC2 and videos are available on the AMOOCON site. Nir’s company is Greenfield Tech.
Nir is also the author of Asterisk Gateway Interface 1.4 and 1.6 Programming
Jerry Shuman of Perssonas (@theagent) joined the call with some great input, too.
Follow on Twitter
@voipusers @e4voip @mjgraves @viperdudeuk @steely_glint @teamforrest @fredposner @asteriskbot
Skype for Asterisk to ZipDX
Talkshoe:
Audio clip: Adobe Flash Player (version 9 or above) is required to play this audio clip. Download the latest version here. You also need to have JavaScript enabled in your browser.
ZipDX/CloudFront:
Audio clip: Adobe Flash Player (version 9 or above) is required to play this audio clip. Download the latest version here. You also need to have JavaScript enabled in your browser.
Today we talked about the EXPERIMENTAL link using Skype for Asterisk (SfA for short). The more we use SfA, the more the Skype people will see that we the Asterisk community are serious about it.
The whole thing began as a phone call made by Tim P. to my Skype number. I almost never have Skype on but the number is forwarded via Gizmo5 to my phone, so when I heard it ring and saw the Skype ID, I realized the call was going through a bunch of servers and trancodings. The quality of that call was striking: total silence when no one was speaking (the equivalent of very black on an LCD display) and very good quality audio with a comfortably small latency. The call lasted at least 30 minutes, too and remaind of good quality.
So, we had at least 6 people called in via Skype from the USA, Israel, Barbados and the UK. The quality varied with the number of channels in use and time, but it was pretty good by and large.In fact, the quality when there was no packets dropped sounded good, even though it was g711.
July 10th: Siemens IP DECT phones
Posted by admin in Future Topics on June 7th, 2009

Our guest is Anthony Stankus, Gigaset Communications product Manager for North America. The Gigaset IP DECT products which include: A580IP system (base and handset), S675IP system (base and handset), A58H accessory handset, S67H accessory handset, SL78H high-end accessory handset.
Michael Graves and I have both talked about how we think these phones are a revolution for the SoHo market. How will they be supported? Who will set them up , provision them, get them working with different pbx and SIP providers?
Tony will be with us on the ZipDX wideband conference bridge to answer these questions and any others you can collect.
More on these phones:
Graves On SOHO VoIP » Newsworthy: Siemens Gigaset S675IP
Alan Lord’s The Open Sourcerer.com
My own modest consumer review
Siemens Phones - voip-info.org
Steven Perich » Blog Archive » Siemens Gigaset IP phones
Sipgate enters the USA Market
Very good call with Thilo Salmon from Sipgate. We’ve been a member for at least 5 years and I was really excited when I got a response on Twitter from none other the the co-founder and CEO of the company.
Audio clip: Adobe Flash Player (version 9 or above) is required to play this audio clip. Download the latest version here. You also need to have JavaScript enabled in your browser.
Open a free Sipgate account and check out what they offer so your questions can be answered and your opinions aired!
Above is the Sipgate user panel. Sipgate offers 200 free minutes to test their system and a free USA phone number. Free calling between Sipgate extensions, of course but you can put several phones on a single account. I did just that and then called myself (from Europe to Europe) and found the lag to be as expected but what a great solution for newbies and people who want to experiment.
Sipgate does voicemail as expected but also, Google Voice-like ringing of multiple extensions. Try it out if you dare.
May 29th: Jim Van Meggelen on Building a solid system without the complexity
Part 1:
Audio clip: Adobe Flash Player (version 9 or above) is required to play this audio clip. Download the latest version here. You also need to have JavaScript enabled in your browser.
Part 2:
Audio clip: Adobe Flash Player (version 9 or above) is required to play this audio clip. Download the latest version here. You also need to have JavaScript enabled in your browser.
Jim Van Meggelen is one of the author’s of O’Reilly’s Asterisk: The Future of Telephony, and an old school PBX guy. Jim is probably a bit of a masochist, which would explain why he got into the telecom business in the first place, and why he now loves Asterisk. Jim is pretty friendly, kinda like a puppy that gets your shoes dirty. His enthusiasm is infectious, but also a little bit frightening if you stand too close. Jim is a partner in Core Telecom Innovations, a recognized leader in the Asterisk Telephony Revolution, and iConverged Inc. He lives in Toronto with his wife and three kids, and loves writing, photography, speaking, improv, choral singing, and old shoes.
Jim will talk about building an embedded-like system using x86 hardware (Intel Atom Mini-ITX) and a stock Linux distro for well under $1000, including FX cards with hardware echo cancellation — this is a great way to build a system that’s almost like an embedded system, but with much less software complexity or incompatibility.
The day the VoIP Industry Rebooted - Jeff Pulver

Jeff Pulver sends a letter out pretty often with a 1998 ALL CAPS PLEASE FEEL FREE TO COPY THIS notice, so here are the relevant paragraphs of his latest missive:
Looking back, I believe May 21, 2009 will be known as a milestone date in the history of the VoIP Industry. For certain it was one of the most significant gatherings of the communications industry in 2009. It was a day that 100+ people from: Across the United States, Israel, Canada, France, Germany and Russia gathered to explore the state of the HD VoIP ecosystem.
Read the rest of this entry »
YATE: Yet Another Telephony Engine
Audio clip: Adobe Flash Player (version 9 or above) is required to play this audio clip. Download the latest version here. You also need to have JavaScript enabled in your browser.
Part 2: Yate and g722
Audio clip: Adobe Flash Player (version 9 or above) is required to play this audio clip. Download the latest version here. You also need to have JavaScript enabled in your browser.
YATE with Diana Cionoiu, who always shoots from the hip. Yate is an open source software developed with reliability, scalability and flexibility in mind for tomorrow’s telephony networks. I guarantee an interesting call this week, but then we always have interesting calls, right? Otherwise, why would they last up to four hours like today’s call?Here is a link to Diana’s presentation at Asterisk Tag 2008. She was also at AMOOCON 2009, here’s the page for her presentation. (Almost all the presentations are now available on the AMOOCON site. Nice work, Stefan!)
Acting upon the suggestion of our group, here is the IRC transcript of the call which again lasted over 4 hours.
VoIP and Social Media AMOOCON Video
Many VoIP geeks aren’t into the social media part of the Internet, yet I find it to be an integral part of both business and pleasure today.

Stefan has posted the video of my presentation at AMOOCON. We all look at our talks and say “OMG, was I really that bad?” I warn you in advance, and have made note of it, my talk starts way too slowly.
The idea was to compare social networking in 1976, 1996 and 2006 and then today, 2009 by evoking my ham radio days in L.A. We used a 220 Mhz repeater and walkie talkies to talk on the phone from anywhere years before the cellphone became ubiquitous. In 1986, I was using the Internet without even knowing what it was with rlogin, rpipe, etc. In 1996, I built a PHP 2 site that basically did exactly what Twitter does, an open IM with a web interface. In 2006, Twitter came along and in 2009 we have a growing number of how to keep track of friends and more importantly, make new connections in our centers of interest.
The link with VoIP is that we now are able to meet people like Dan York, Jeff Pulver and many other movers in the VoIP space through all the usual channels and hopefully through the our own VUC as well. The VUC started around March 2007 and has grown in many ways since then, thanks to Mark Spencer’s immediate support and later, Digium’s realization of the importance of the whole social transformation that is taking place on the Internet. We have had some help from the people whose logos you see at the top of the page and I’m grateful to them and of course to every participant, especially the core group of “friends”, all of whom I hope to meet some day.
r/z








