AMOOCON 2010 will, as always, be interesting and fun in and of itself. It’s great to meet people in person and talk about what they’re doing.

We’ll be talking to the people who offer Blink, a free OS X g722-capable SIP client (and much more) and longtime VUC regular Tim Panton, who among other things is the reason we can connect to the VoIP Users Conference  via Skype. The Skype bridge is provided by PhoneFromHere.com.

Blink is offered by AG Projects. Read more info about Blink. or download it here. AG Projects CEO, Adrian Georgescu was our guest in previous sessions. Phone from Here’s Tim Panton was recently involved in the “Telephony Island Adventure“.

Freetalk Connect wants to put Skype on every phone in your office.

Intelligent routing capabilities:  incoming Skype calls, as well as calls over SIP, the PSTN and IAX2, can be routed by the FREETALK Connect to any local or remote Skype user, SIP endpoint, analog or mobile phone.

Set up is simplified, enabling small business users that are not tech savvy to use it quickly and simply, without formal training. Supported telephones are plugged into the company’s network and the device auto-detects and configures them. An on-screen wizard then guides the customer through a few business-related questions that help configure the company’s communication system and enables Skype calling from every supported desktop phone in the office. Adding users and administering the system after installation is just as easy.

We’ve been testing Skype for SIP open beta today. This allows you to connect Skype to your commercial PBX, much as Skype for Asterisk allows connection between Skype and Asterisk. Listen to the recording to learn more about the experiences to  date.

In addition to all of the Orgasmatron magic including free calling in the U.S. and Canada courtesy of Google Voice, you now get some terrific new features: randomly generated passwords for all of your extensions, free Skype support and a new backup module both of which we’ll introduce over the next few weeks. And CallerID Superfecta now is preconfigured to work out of the box with support from dozens of providers worldwide.

Among the participants Tim @steely_glint, Neil from IdeaSIP, Karl @karlfife, Michael @mjgraves and Graves on SoHo VoIP, David Frankel from ZipDX who furnish our great wideband conference facilities, Dave @DaveMichels from Pin Drop Soup, Pascal from Media5.

Welcome to a very old friend of mine, Bob from Future Video. The VUC encourages new people to be a part of our weekly talks. Please join us so you can chime in, ask questions and generally have a great time.

Thanks to Digium andfor allocating more channels of Skype for Asterisk to reach the ZipDX conference bridge and to John Todd for making this happen! We had a lot of live callers, including many on Skype.

Unixdawg on FusionPbx
Download audio file (TS-288078.mp3)

Part 1 Download audio file (TS-256280.mp3)
Part 2 Download audio file (TS-264537.mp3)

Lots of talk about Skype for Asterisk, with Tim Panton who has been on the SfA beta for over a year and who will be speaking at Astricon in October.

IRC Transcript 2009-09-04

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.

IRC Text Transcript 2009-06-12

IRC Transcript 2009-04-03

I’ve been testing all the ways I can find to connect between phones and Skype. I happened to be testing OpenSky when I saw some posts from Michael Robertson on the asterisk-users mailing list. As I exchanged a few emails with Michael, I realized he had been a founder of Mp3.com and well as Gizmo5.

He was generous enough with his time to spend a full hour (on a cellphone!) answering questions and sharing thoughts about the various SIP to Skype solutions, including Digium’s Skype for Asterisk, Gizmo’s OpenSky and Skype’s SIP for Skype which Michael calls vaporware.

Post conference chat, aka “Cognac”

The post conference has discussions about the expectation that all services should be free with David Frankel of ZipDX.com Randal Schwartz from FLOSS Weekly and the regular gang. Due to some problems with Talkshoe, various audio sources are used so you can hear what different phones sound like.

Download audio file (TS-210205.mp3)

Download audio file (TS-170693.mp3)

Skype for Asterisk with Steve Sokol, Digium product manager. Also a general talk about life, the universe and everything such as the high points of VoIP in 2008.

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