Follow the contest on Google Plus
When Gerd Graumann last visited with us in December, we talked about Speech Technology Group‘s  natural text-to-speech voices and launched the speech humor contest. In this session he will announce the winner and play the funniest submissions.

The discussion will be on speech recognition technology, which has lately taken center stage though the widespread mobile adoption. Gerd will share with us how Speech Technology Group has taken a powerful 64-bit ASR that is used by thousands of companies exclusively within the unified communications space to work now with every platform that supports the industry standard MRCP interface.

Kevin Fleming and Malcolm Davenport call in from Europe for a session on the new Digium phones announced a few days ago.

More about the phones from Dave Michels on No Jitter

We will be testing the HarQen Symposia product in this session, with Marc della Torre. Symposia is a hard thing to describe in a few words, you really need to see it to understand what it might bring to a conference. This is exactly what we hope to see happen for this conference.

Recording with bookmarks

Key moments can be marked by participants (see the timeline of the recording below).

 Views of finished Symposia recordings are available and can be shared via email links. The capture on the left shows a Participants list, a Search facility, Flags such as “concern”, “to-do”, etc, Topics and Mentions in the left hand column of that screen. On the right hand side, Shared Moments, Starred mMoments, and Images Shared are visible.

There are some great concepts here for a focused business conference that make HarQen’s quote understandable: “Ever sat through an hour-long webinar or meeting and afterward wished you could share the two minutes that mattered with someone else?” I believe you’ll have to see this to understand it, so we’ll try to coordinate all of our our usual channels, ZipDX, IRC  with this moment and make sure everyone who wishes can participate.

The SIP school offers training and certification in SIP and other VoIP technologies. Use VUC discount code VUC10 for 10% off!

As the provider of the global standard in SIP training and certification, The SIP School has taught thousands of employees in the telecommunications industry how to better  support their clients, products, and services. Until recently, students training to become an SIP School  Certified Associate (SSCA®) were instructed in their first session to create a SIP address with any free service. Today, The SIP School announces another option by working with OnSIP as their SIP  service provider – leveraging the OnSIP API to provision each student with a SIP address on thesipschool.com domain.

This is the original press release.

The recorded session begins with a reminder of the TTS jokes contest where your entry might win a Kindle Fire. Winner announced Friday, February 3rd on the VUC.

Olle Johansson joins us again in a rare visit to discuss the state of IPv6 adoption. Also heard from, lost of us who have or have not done it and why or why not.

We did a VUC Google Plus Hangout during the call as we will often be doing. These usually begin an hour or so before the actual conference. Please make sure you have the VUC G+ Page circled! During the hangout we looked at the Invoxia Desk Phone for iOS devices and the Polycom VVX-500 touch screen SIP phone.

The recording is an informal chat (is there any other kind of VUC?) about Republic Wireless (why won’t Brian talk to us?) and Invoxia’s iOS desk phone, the one Michael Graves is evaluating.

2011 was a fantastic year for VUC. I was able to get to Astricon, thanks to Digium, and met several new VUC folks (such as the ubiquituous James Body) as well as seeing old friends again, including the elusive John Todd 2.0.

We’re on Google Plus this year and we hope to see it grow: http://gplus.to/vuc. Please spread the word. G+ is a great way for us to communicate about each week’s sessions. Don’t be shy, that’s what we’re there for.

We had help from great sponsors and partners: Voxeo Labs, Sangoma, OnSIP.com, e4strategies and of course our great partnerships with ZipDX and Voxbone.

We chatted with amazing people doing amazing things: Allison Smith, Bryan Johns, Simon Wood, John “Captain Crunch” Draper, Kevin Fleming, Chris Veazy, Danielle Morrill, Chris Matthieu, Diana Cionoiu, Gerd Graumann, Jason Goecke, Dan York , David Burgess, Steven Song, Dave Michels, Ward Mundy, Brough Turner, Eric Klein, Nir Simionovich, Ben Klang, Alok Saboo, Alex Balashov, Andy Abramson, Bryan Martin, John Riordan, Charlotte Oliver, Emil Ivov, Rob Wolpov, Mike Oeth, Thilo Salmon, Frederic Dickey, Pascal Doré, Todd Carothers, Randal Schwartz, Jim Gettys, Dave Taht, Hugh Goldstein, Adrian Georgescu, Andrew Kuan and all the great regulars!

Joining us this Friday is Vidtel, a cloud-based video conferencing service that is easy to use and price-disruptive based upon Any to Any model. Spanning boardrooms, mobile workers and everything in between, the Vidtel Meet Me service allows room-based video conferencing systems, executive desktop video systems, PCs/Macs, smartphones and tablets to work together. Any combination of SIP, H.323, Skype and Google Talk technologies can interoperate on demand with Vidtel’s service. The cloud service is targeted to the small-medium enterprise market, priced for any-sized budget and is as easy to use as a telephone.

Scott Wharton, Founder and CEO of Vidtel, is a recognized leader in the VoIP and video industry and will join the call to discuss video landscaping and why any-to-any video calling will be important in the coming years.

From 1999 to early 2008, he was the VP of Marketing at BroadSoft, Inc., the leading developer and marketer of VoIP application servers, which also handle video IP calling. Scott made significant and enduring contributions leading to: the dominance of the BroadSoft brand in the voice application space, the company’s acquisition of 9 of the world’s 10 largest telecom service providers as customers, and the expansion of the company’s presence in over 60 countries. Scott also defined a new product category for the telecom industry, a next- generation applications layer, establishing BroadSoft’s global leadership in the VoIP applications market.
In 2003, Scott was honored with the Frost & Sullivan IP Telephony Executive of the Year Award. It is presented each year to the individual who has excelled in developing and executing a business strategy allowing his or her company to gain market leadership. In 2006, Scott was recognized as a “Top 100 Voice” in the IP Communications Industry by Internet Telephony magazine.
Prior to BroadSoft, Scott was Vice President of Marketing at VocalTec Communications, the pioneer and a leading developer of VoIP software solutions. From 1995 to 1999, Scott spearheaded numerous groundbreaking initiatives, including the development of the first commercial VoIP product and the creation of the VoIP service provider industry. In the 1990s, Scott created one of the first IP video products for the consumer market as a senior marketing exec with VocalTec, the pioneering VoIP company. Scott holds an MBA from Yale University.

HarQen, the company that unlocks the value of the human voice in business conversations, joins us to discuss Symposia – the end of the dead-end conference call.

Symposia is a cloud–based collaboration offering that dynamically transforms any conversation into an interactive synopsis that lives long after the conference call, meeting, webinar or presentation has ended. The use cases are endless from CRM to recruiting.

HarQen’s VP of Business Development, Marc Della Torre, will be joining the call to show how Symposia lets users better take part in calls and do more with the call when it’s over than ever before.

Prior to joining HarQen, Marc lead sales and business development for JAJAH from late 2006. Marc transformed JAJAH’s free consumer VoIP telephony service into a profitable carrier-friendly cloud telephony service generating over $30m annually, which resulted in the $207m acquisition by Telefonica in January 2010. Prior to JAJAH, Marc lead the VeriSign mobile merchant services team which created the value which PayPal pursued via the 2005 acquisition for $370m. Marc received his BSEE from Wayne State University and a MSEE from George Washington University.

Our guest is Gerd Graumann, President of Speech Technology Group,  who provide advanced TTS and ASR speech technology.

Using Text-to-Speech online, site visitors can now listen to anything you have posted on your website. With a simple click of a button, your content is instantly converted to audio and played back. This is but one of many applications, where the new generation of natural TTS voices can help. IVR is another example, where TTS plays a significant role. A high quality TTS voice combined with an affordable license model not only sounds good, but can be a significant contributor to the success of a project.

Examples of these highly natural voices are at www.speechtechnologygroup.com

The speech synthesizer can be integrated using a wide variety of interface options, such as MRCP, API or Web services, whether you are looking to deploy an embedded, mobile, desktop, server or cloud based solution.

In this session, we will be discussing improvements in TTS technology and  launch an unusual contest that should make you smile.

Michael Cullen of Criticall 911 joins the call to discuss their real world application of this technology.

As shown by the recent arrest in the case of terrorist who were hacking AT&T business customers to fund Al-Qaeda; Telecom Fraud has come a long way from Captain Crunch and Steve Jobs phreaking Ma Bell for fun and glory. It is now big business aimed at stealing from you via your PBX. Let’s discuss actual cases and some things you can do to make sure your PBX is not funding terror.

Eric Klein of Humbug Labs will attack this topic on short notice, due to a cancellation. Here’s a link to the Humbug Blog.

Add your questions and comments for Simwood and ThreatSTOP to the VUC G+ post

Simwood provides international carrier services with wholesale IP telephony and a portfolio of managed services, including their pioneering use of  ThreatSTOP in the VoIP space (and also DDoS mitigation).

ThreatSTOP is included in the Simwood DDoS Security solution, as the second layer in multiple tiers of defence. It is also available as a product in its own right. It can represent a first step to full protection by complementing existing security solutions, but it can also complement the full Simwood solution by offering an additional layer of bespoke on-site protection. Notably, the service offers the option to block by country, maintain customer-specific block lists and include/exclude sub-lists to reflect the level of tolerance desired.

We talked a little about Republic Wireless, but even those who have signed up will know nothing before mid-December. Moved on the Google Voice, wondering about whether it will remain free in 2012 or what the plan is. We’ll know soon, we guess.

Surprise and brief visit from John Draper (Captain Crunch), chat about the problems of security audits.

Missed James Body who could have told us more about some of the mixed wi-fi and 3G mobile schemes.

Shilled again for Google Plus and in particular the VUC Page on it. If more participans could annote the weekly posts on that page, the VUC would become a better organized resource for all .

If you are new to the idea of VUC, nothing is easier than calling free via Skype, thanks to Phone from Here.com. On Windows, you can just type it into the dialpad like so:

We did discover that in the latest OS X version of Skype, you can not type a dot in names (like vuc.me). Bummer!

]]> http://www.voipusersconference.org/2011/open-discussion-on-voip-and-mobile/feed/ 5 Captain Crunch,Google Voice,James Body,John Draper,Mobile networks,Mobile VoIP,Republic Wireless We talked a little about Republic Wireless, but even those who have signed up will know nothing before mid-December. Moved on the Google Voice, wondering about whether it will remain free in 2012 or what the plan is. We'll know soon, we guess. - We talked a little about Republic Wireless, but even those who have signed up will know nothing before mid-December. Moved on the Google Voice, wondering about whether it will remain free in 2012 or what the plan is. We'll know soon, we guess. Surprise and brief visit from John Draper (Captain Crunch) (http://savingcaptaincrunch.com), chat about the problems of security audits. Missed James Body who could have told us more about some of the mixed wi-fi and 3G mobile schemes. Shilled again for Google Plus and in particular the VUC Page (http://gplus.to/vuc) on it. If more participans could annote the weekly posts on that page, the VUC would become a better organized resource for all . If you are new to the idea of VUC, nothing is easier than calling free via Skype, thanks to Phone from Here.com (http://phonefromhere.com). On Windows, you can just type it into the dialpad like so: (http://www.voipusersconference.org/wp-content/uploads/2011/11/VUCskype.png)We did discover that in the latest OS X version of Skype, you can not type a dot in names (like vuc.me). Bummer! @voipusers clean 2:17:22 http://www.voipusersconference.org/2011/vuc-google-plus/ http://www.voipusersconference.org/2011/vuc-google-plus/#comments Tue, 08 Nov 2011 09:20:25 +0000 VUC http://www.voipusersconference.org/?p=3585

The brand new VUC page on Google Plus:  http://gplus.to/vuc
All we need now is YOU

OnSIP Hosted PBX offers free SIP hosting and telephony services to small and medium sized businesses.  Voxeo’s Tropo is a cloud-based platform for web developers to build voice applications.  What happens when you bring these two together?  Find out on the next VUC, when John Riordan and Rob Wolpov, founders of OnSIP Hosted PBX, and Chris Matthieu, Director of Business Development at Voxeo, join to discuss a recent OnSIP-Tropo application and what’s next.

In the photo below, Nicole, John, & Jonathan Taylor ( Voxeo CEO).

A Quick look (full details in Nicole’s Blog)

Rooted heavily in SIP standards and customer service values, Voxeo has been a company we’ve known about and respected for years. Since it was founded in 1999, Voxeo has been mainly focused on hosted and on-premise based IVR solutions for customers much larger than ours. Likewise, our business rarely intersected with Voxeo’s. However, in 2009, Voxeo launched a new product called Tropo, a cloud-based application platform that makes it easy for web developers to build voice applications using familiar languages. This product is extremely promising for us, and it also gave us a great excuse to head down to Orlando for Voxeo’s Customer Summit.

The application then works as follows: When an inbound call comes in, it is placed in a conference. Before putting the call in a conference, this same script is launched via the Tropo API which then makes an outbound call to the callee. If the callee answers, the callee is prompted to accept the call. If the callee accepts, the callee is also put in the conference, thus ‘bridging’ the two call legs. If the callee declines or does not answer, the caller is removed from the conference by yet another call to the Tropo REST API to send an event that signals the caller exit the conference. At that point, the caller is sent to voicemail if a failover address is provided; otherwise, the caller is hung up on.

Here’s a post from my personal account on G+.

There are already too many channels, but I can’t find a channel that everyone is listening to, and it’s a problem. I will cross post this on at least 5 of them: email, Facebook, Google Plus, IRC, and the Unblog.

Photo: vintageham.com

While at Astricon, every single night there were ad hoc groups forming to go to dinner. Some had American cells, some got the wifi cut off and had no data, etc. So how were we to keep in touch? The irony of a large conference entirely about telecommunications, convergence, VoIP and telephony is surely not lost on you, gentle reader. We really need a single channel if we want to keep in touch. That channel, for my part and until something better comes along is Google Plus.

If you want to be included in future dinners, wine tastings, lunches, breakfasts, strip clubs, poetry readings or whatever comes up when we get together in meat space or virtually, please get on Google Plus. It matters not if you do not follow anyone or are not active, but it does matter that that channel is the most flexible at the moment. Adding my prejudices, Facebook isn’t good for this, IRC is too old school, email too iffy. Assuming you have a cellphone and data, Google Plus is the way to go. If you eschew social networks – of which Google in NOT in my opinion – fine. But don’t be surprised if you don’t hear about things in a timely fashion.

Be sure to look at our Live Blog for photos, videos and interviews from Astricon!

How can I list the names of everyone I met and enjoyed hanging with? Here are a few but I know I’m forgetting many: Allison, Andrew, Ant, Barry, Ben, Dave, David, Eric, Evan, Jacinda, John, Julie, Mark x 3, Michael x 2, Nir, Rebecca,  Tristan…

Would you buy a used PBX from these guys?

Kevin Fleming  joins us to talk about Digium‘s Redundancy Series products.

Redundancy Series (aka “R-Series”) products, are a family of inexpensive, versatile appliances that enable physical layer failover of analog (POTS) and digital (T1, E1, PRI or BRI) telephony connections to Asterisk-based communications systems. The R-Series devices allows Asterisk administrators to create high-availability solutions that insure uninterrupted business communications, even in the event of a catastrophic hardware or software failure. R-Series appliances include all the tools necessary to build a fully redundant active/passive communications solution. If the primary server experiences a hardware or software fault, all telecom services will automatically switch to the backup server.

The R-Series integrates with software to mirror file systems, manages the transition of a shared IP address and monitors system health between the primary and backup. The R-Series family consists of 2 models, the analog R800 appliance and the digital R850 appliance. Each model supports eight (8) public switched telephone (PSTN) circuits. The appliances deliver PSTN circuits to each Asterisk server via Digium’s telephony interface cards. Management and failover communications between the Asterisk Servers and the R-Series appliance is done using a set of USB connections, which also provide power—saving energy costs over the lifespan of the appliance.

Allison is back with the latest news of her activities in the world of voice-over and voice prompts. If you’re not familiar with the name Allison Smith, I’ll tell you when I first heard it: on Free World Dialup circa 2003, yes, nearly 9 years ago!

If you’re lucky enough to be coming to Astricon this year, you’ll certainly have a chance to meet Allison, one of the nicest people I’ve met anywhere.

PBX in a Flash from Nerd Vittles, Tim Panton on using Open BTS at Burning Man and a bunch of other interesting stuff that comes up before we have film of it!

Links (Open BTS / Burning Man):

http://www.disruptivetelephony.com/2011/08/video-how-to-communicate-at-burning-man-using-openbts-and-tropo.html

http://openbts.blogspot.com/2011/09/burning-man-2011-yes-we-were-there.html

Meet the Invoxia NVX 6210, a desktop phone that integrates the iPod, iPhone or iPad as a control interface. You make calls by dialing from your iPod, iPhone or iPad using the dedicated invoxia app. You can now receive calls made to your mobile as well as calls to your office on the Invoxia phone. To answer a call in handsfree mode, whether it’s coming from your mobile or desktop phone, simply press the volume button. There are just two sensory buttons on the invoxia phone’s base: one for muting the mic, and the other to access your voice mail.

Features

A wide range of connections are available on the invoxia desktop phone:

• 2 USB ports (for the corded handset, your iPad and for headphones, for example)
• 2 Ethernet ports, including 1 PoE port to power your phone and to connect your network cable and computer.
• iPhone/iPod docking station
• Bluetooth technology for wireless communication with your iPhone or iPod, or to pair your phone with a wireless headset (Headset Profile or A2DP),
• DECT technology for optional cordless handsets

Radvision develops products and technologies for unified visual communications over IP and 3G networks by combining video, voice, data and wireless – in room Video Conference Systems, mobile video solutions and video-enabled desktop platforms for IP and emerging next-generation networks.

Here is a brief intro to our guests from Radvision this week:

Roger Wallman is director of product marketing at RADVISION. He has more than 20 years of experience in video conferencing, having started his career at PictureTel in the early 90s. Roger has also held key marketing and product management positions with VideoServer, CUseeMe Networks and First Virtual Communications. Roger graduated from the Wisconsin School of Electronics and the Milwaukee School of Engineering.

Robin Raulf-Sager is director of communications at RADVISION, which she joined in May of this year. Prior to RADVISION, Robin spent a number of years at tech companies including Polycom, SpectraLink and StorageTek. Robin also currently serves as the marketing work group chair for the UCI Forum. Robin has a master’s and bachelor’s degree in communications from the University of Illinois.

Don’t forget about Astricon, several of us will be there from VUC, we’ll be able to hang out, isn’t that what it’s all about?

One of the slides from Jose’s presentation:

Guest : José de Castro

Published July, 2011

What could be better than a new book by Dan York, to get you on the right track for IPv6 adoption?

Today’s guest Blinkmind – more to come after the session.

Win a free Blink Pro license for OS X by getting on Twitter and watching and using the #icanblink tag. A question will be asked at the beginning of today’s session. Answer on Twitter with the tag #icanblink and you may win. 4 copies to give away, instant download; If you do not use a Mac, you can give the license away to someone who does.

Our friend Emil Ivov checked in from Bulgaria to share the progress of the Jitsi project. We’ve been following the project since it was called SIP Communicator. The more recent April 2011 discussion is here.  Here are some of the subjects covered.

* Jitsi recently added support for Audio and Video Calls with GTalk
* Differences between standard XMPP/Jingle and Google’s variant (aka Gingle)
* Incompatibility with Google’s telephony mechanisms
* Google’s pseudo TCP
* Ongoing in Jitsi: an Android port.
* Security, encryption
* multi-party conferencing

Adhearsion is Telephony through Ruby-colored lenses.

Ben Klang tells me, “In the last year the project has made great strides, reaching version 1.0 and going well beyond.” We’ve heard from Adhearsion in the past (link from 2010).

Come hear an update from the Adhearsion community and discuss some of the exciting advances that have been made. Ben and others will also share details about their upcoming conference, AdhearsionConf, held in October in San Francisco.

Twilio provides infrastructure APIs for businesses to build scalable, reliable voice and text messaging apps.

The Twilio Conference happens Wednesday, September 21, 2011 at 8:00 AM – Thursday, September 22, 2011 at 6:00 PM (Pacific Time) in San Francisco. Here’s a discount offer.

You may recall we talked about Twilio’s OpenVBX almost exactly one year ago.

The death of WebOS hasn’t been greatly exaggerated, apparently. Some very interesting discussion comes out of this topic and some Google-Motorola trash talk as well. Batteries not included, but the usual endless conjecturing is.

Dan York brings some info from SpeechTEC. The mobile use of speech recognition is a specific focus we talk about. Also, the problem of the cloud and national data compliance, as in Europe, where data is not supposed to cross borders.

Lazboy and a few others give us a rundown about ClueCon.

Talk about Chrome Beta Audio without Flash or Java.

MRCP

We’re also using Google Plus to post threads you can comment on live. This leaves a nice permanent record.

Someone needs to make notes of what this is about!

We will be giving away 1 OBi110 and 1 OBi100 to “lucky” listeners. See posts on Google Plus.

If you don’t win one, you can order an OBi100 VoIP Telephone Adapter and Voice Service Bridge from Amazon, which will also throw a buck or two into the VUC beer fund.

Obihai is just getting going, but are already enjoying the popularity the OBi has received from end-users, VARs and VoIP SPs. We will talk about what they’ve done in the SIP, XMPP, P2P and social networking technology space and how this is related to the explosion in social networking apps, the popularity of VoIP services and the ‘un’ popularity of the land line service.

They sent me an OBi110 evaluation unit. I connected and plugged in the lightweight, white plastic box, called the echo test number, and it worked perfectly, with a low lag time from here in Southern California. Fine, but for us logtime VoIP users, not exactly a revelation. However, the ease of connecting to a Google Voice number suddenly makes this very reasonably-priced ATI very interesting.

VoIP scams and fraud: the mildly bothersome, the ethically dubious, and the outright illegal of business models used by unscrupulous VoIP service providers (and other elements of the supply chain) to game the system in order to sell you (and each other) lemons.

In the aftermath of the big and controversial voicemail hack starring the NOTW… wait – “controversial”, does anyone think this was OK? Anyway it’s our topic and we hope to gather some knowledgable people to chat about it including Tim Panton, Humbug Telecom’s Boaz Bechar, Truphone’s James Body, Dan York of Voxeo and anyone else ready to “testify”. Be there!

Video of the talking heads is here

Watch live video from voipusers on Justin.tv

http://gplus.to/vuc

Lots of interesting features separate this from the usual large social networks. The first one is hangouts, ad hoc video conferences between 2 to 10 people. They work very well, have low lag (hi Karl) and are available with a single click. (Well, then you do have to maybe check your audio and video settings). Hangouts are both fun and functional. So much has been written about G+, we do not need to go into a tutorial here. I (randulo zeeek) just wanted to give you our take on it, I think it’s great and that you should try it and see whether it works for you.
The idea is to look at who “VoIPusers Conference” has in its circles because it is only following people who have some connection with VoIP, telephony, etc. If you are on G+ and not in a VUC circle, let us know by sharing a post with +VoipUsers Conference. You can do this privately on G+ by sharing with just one name:

You can then see who is in VoipUsers circles by going to the Profile Page

Important note: Before you write to VoIPusers Conference on G+, please make sure you have a profile with some info in the about section. Of special interest to us is that we look to see if there’s a VoIP element to what you do or in your interests. We also look at your posts and see what you’re talking about. Make it easy for us to tell you’re into voip and hey, why not mention the VoIP Users Conference or VoIP & Tell if you regularly participate, even as a listener?

The usual suspects gather to discuss a myriad of issues, technologies and solutions around telephony, VoIP and networks. You should join in, your voice would be appreciated. This is an open discussion, anyone is welcome to call in and chat, ask questions, contribute answers or tell geeky jokes. That’s what we’ve been doing every week since March 2007, over four years ago.

Join the VUC at Astricon 2011, we have a discount for you.

Give us a +1 for Google above!

eComm 2011 VUC Specials by VoIP Users Conference

Andy Abramson, Dan York and whoever else is able to function coherently (or not) at 7:45 AM USA Pacific Daylight Time will be on VUC in three special live breakfast sessions from eComm2011: Emerging Communications Conference, next week. Speakers at eComm are some of the brightest lights in VoIP, including Tim Panton, Dan York, Darren Schreiber, Johnny Diggz, Voxeo CEO Jonathan Taylor, Lee S Dryburgh, Digium’s Bryan M. Johns and many more.

Tim Panton, Jason Goecke, David Frankel and Karl Fife on this final session. We talk about Skype’s plans to abandon Skype for Asterisk, Tim gives a preview of his talk, more about what’s coming in the standard-driven areas of VoIP and more.

Andy Abramson, Dan York, David Frankel, Jim O’Brien (Counterpath Director NCG Customer Engineering), Kelly Fitzsimmons, Karl Fife and more on day two of eComm 2011 in San Francisco.

Andy Abramson, Dan York and Tim Panton are joined by Alec Saunders, Erik Lagerway, Martin Geddes, Todd Carothers, “disruptive” Dean Bubley, Aaron Huslage and (more luminaries coming tomorrow) for a round table on the VoIP of tomorrow. This who’s who of VoIP is too hot to miss. No handsets were harmed in the recording of this episode.

Register here for the Sangoma rebate program, click on the logo below

Our open mic session was on Michael Graves’ new Sennheiser headset and Truphone’s James Body was the surprise guest. Also talked about Authorize.net, what assholes they are and their policy on recording calls but threaening to hang up when you ask to record a call.

Just as I went to my banking site to lodge a complaint about these issues, Authorize.net has issued a full refund. I thank them for listening to reason and I hope this incident helps them deal with the public in a more open way.

Having opened an Authorize.net account in order to collect payments from overseas customers without sending checks back and forth over the Atlantic, I was asked to a pay a $120 setup charge and was immediately billed for the monthly gateway fee of $20. While I was attempting to set up a merchant processor, a month went by and I was billed another $20. It turned out to not be possible to get a merchat account because of my lack of a credit history, so the Authorize.net account is useless to me.

I called Authorize.net and spoke to a few sales people about a refund. I wanted to record the call, but after the person I spoke to said (live) “we may record this call for quality assurance” I said I also am recording it. She then said, I will have to disconnect, then.

Authorize.net provided absolutely no service to me during the five weeks I had this account open. At first they claimed there were transactions on it, but as I told them that was me doing one test. I could have left a trial account open for a bit. The trial account is free. The point is that in good faith, Authorize.net could simply fully refund my money as I will not be able to use their service. They are claiming that it is not their policy to refund. I say, “refund in good faith” in this case, your “policy” shouldn’t apply. There is no reason Authorize.net should be keeping money they didn’t work to earn, is there?

With SMSified, you can build nearly any SMS application you can imagine. Auto respond to inbound messages, send interactive outbound notifications, drive sales with promotional offers, enable two-way customer self-service applications and much more. Chris Matthieu and Dan York join to chat about how SMSified works and what it can do for you.

Lots of great examples here on the SMSified Blog.

Sipgate CEO Thilo Salmon returns to VUC this week with Diana Cionoiu, to tell us of their use of Yate. We see on the Yate site that extensive use has been made of Sangoma hardware products.

Press release:

The open-source telephony engine Yate is making inroads into what used to be the domain of proprietary equipment provided by large telecom suppliers. Following the certification of Yate SS7 stack by Deutsche Telekom’s test lab, Sipgate has started to deploy Yate at the core of it’s softswitch. The decision came following extensive tests to determine stability, scalability as well as resilience.

“The open source telephony platform was easily integrated in the infrastructure of our softswitch”, said Thilo Salmon, CEO of Sipgate. “We are mostly impressed by the quality and scalability of Yate technology.”

“We are pleased that sipgate has chosen Yate for implementing their telephony softswitch”, said Diana Cionoiu, managing director of NullTeam. “This is a solid recognition of our 7-years development of Yate platform”

Part 2
After Hours 326: Death of the Desk Phone by VoIP Users Conference

8×8, Inc, recently introduced a new web-based UC offering called 8×8 Virtual Office Solo, designed specifically to meet the needs of single person businesses and is disruptively priced at $7.99/month. Bryan Martin, 8×8 Chairman & CEO, will provide an overview of the service and discuss the strategy behind its launch.

Virtual Office Solo includes an iPhone app, built-in faxing and a beta social media section for Twitter and Facebook connections. More about  Virtual Office Solo product on GigaOm Webworker Daily. Article: All VoIP News.

We welcome John Riordan, CTO, and Charlotte Oliver, Systems Engineer, from Junction Networks for this talk. IPv6 has serious benefits for SIP, so Junction Networks has made it a priority to begin making their network and systems IPv6 ready. Currently, they are directly connected to the IPv6 Internet via multiple providers and have established network infrastructure and redundancy. Also on the topic of OnSIP platform preparations for the future, Junction Networks will discuss their patent-pending process for scaling SIP proxies., including their recent milestones in establishing an n+1 architecture, and how this supports their end goal to deliver high quality service to OnSIP customers.

Frederic Dickey, in charge of Sangoma Product Management, joins the VUC today to talk about Netborder v4.0. NetBorder Express software, together with Sangoma’s computer telephony boards, is a complete SIP-compliant VoIP Media Gateway solution, delivering the most flexible and cost effective solution by removing the need for external gateways. Compatible with Analog FXO and FXS, MFC/R2, ISDN BRI and PRI telephony interfaces, NetBorder Express can scale from 2 to 480 simultaneous calls, providing excellent value and flexibility for deployments in any geography.

On this session, Pascal Doré of Media5 Corp joins us to talk about the evolution of their products since his last visit over one year ago. Media5 Corp’s products include SIP clients for IOS and coming soon, for Android.

Dave Michels (@DaveMichels on Twitter) will finally clue us in on Lync, a topic he’s covered many times in his excellent blog that you can follow at www.pindropsoup.com.

Microsoft’s VoIP solution known as Lync is changing the conversation at the enterprise. Rarely does a single product cause so much controversy and impact. Dave Michels will provide an overview of Lync, and an analyst’s view of why its stirring the pot.

Todd Carothers, Counterpath’s senior vice president, marketing and products, will demonstrate the recent release of Bria 3.2. Todd will review new features as well as discuss how this release sets up the foundation for release 4.0. Todd will also present Counterpath’s Mobility product suite,  specifically Bria Android Edition updates, Bria iPhone Edition 2.0 and Bria iPad Edition 1.0.

Bria is a highly secure, standards-based, multi-platform softphone that enables voice and high-definition video calls, making it ideal for enterprises, government agencies and other organizations that want to replace their desk phones or add unified communications functionality to their existing IP phone platform.

New to Bria 3.2

* Multiple Account Integration
o Communicate with contacts from different sources and accounts, including local or company directories, Microsoft Outlook, XMPP, XCap or WebDav servers.
o Merge contacts into a single view with all of their information from different sources in the same place.
o View a contact’s presence status on their various accounts in order to choose the communication method to reach them best.
* Bria Add-in for Outlook® (Windows Only)
o Make calls directly from Microsoft Outlook® emails and contacts
o View missed calls and new voicemail notifications directly in Microsoft Outlook®
o Control main Bria functions directly in Microsoft Outlook®
* Company Chat Rooms
o Join company chat rooms for conferences and meetings
* Improved Contact Management, Display, Sound Controls (Volume and Mute) and Workgroup functions

From Wikipedia:

Extensible Messaging and Presence Protocol (XMPP) is an open-standard communications protocol for message-oriented middleware based on XML (Extensible Markup Language). The protocol was originally named Jabber, and was developed by the Jabber open-source community in 1999 for, originally, near-real-time, extensible instant messaging (IM), presence information, and contact list maintenance. Designed to be extensible, the protocol today also finds application in VoIP and file transfer signaling.

Emil Ivov of Jitsi.org (formerly SIP Communicator) and Thiago Rocha Camargo (of Nimbuzz) join us for a day of Jabber. Here’s a rough outline of what we’ll be talking about.

* What is XMPP/Jabber
* How does one do telephony with XMPP
* How does XMPP/Jingle compare to SIP and (why) is it better.
* Who supports it
* Facebook and their XMPP gateway
* Google Talk
* Nimbuzz – one of the biggest VoIP providers using XMPP as their primary protocol
* NAT traversal
* How does one do it with XMPP
* Again, how is this part different from what we have with SIP
* Media relaying with TURN and Jingle Nodes

And so much more… two hours of it!

Mike Oeth of Junction Networks called in to tell us about their newest development, “Get On SIP”. We also spoke of the recent EC2 outage and some ITSP-related issues such as proxies and SIP registration.

getonSIP is a fast, easy and FREE way to register a SIP account, allowing you to communicate using audio and video over the Internet using the SIP protocol. getonSIP uses the same technology as Junction Networks’  hosted enterprise phone system, OnSIP, so you get plenty of great features, including HD audio, voicemail to email, and much more.

http://www.getonsip.com

Chris Matthieu on Tropo.com, great added comments by Tim Panton. Michael Graves, just back from NAB with a few notes. Karl Fife on WebOS (just kidding!)

Bad Wifi at the trade show, new Siemens Gigaset handsets and more DECT comments. Profiles on mobile devices, sharing books on Kindle, Nook, Android, iOS, etc.

Matt Bramson of InPhonex  joins us. To enhance collaboration and communications for small and medium-sized businesses, InPhonex recently launched Televate, a cloud-based telephony system that combines the business-communications capabilities of a hosted PBX, hosted IVR and hosted CRM. Televate, to be sold through channel partners, is designed to help SMBs appear larger than they actually are and ramp up productivity to compete more effectively. The service works anywhere inside or outside the office, enabling SMBs to support global operations and customer bases. Unlike other cloud-based telephony solutions, Televate empowers every employee with full call-center functionality, including CRM integration, content-rich screen pops, and caller and customer history.

The Televate CRM application will suck in contacts from a variety of sources — including Outlook, Google, LinkedIn, and Facebook — into a single management interface. On the telephony side, Televate’s feature set covers the typical PBX list: voice, voice mail, email delivery, auto-receptionist, local and toll-free numbers, fax, and virtual extensions. IVR functionality supports a variety of automated call management features.

InPhonex places heavy emphasis on mobile integration, too, noting that smartphones have added fuel to a common phenomenon in smaller businesses: Everyone, from owner to intern, is a customer service rep. Televate launched with an Android app that offers the full functionality of its desktop UI. The company will release iPhone and Blackberry apps in the second quarter.

Part 2, a wandering discussion:

0:04:40 webOS
0:08:32 Karl moves to the Jabra headset
0:13:09 Sokris 5501
0:16:30 MobileMe
0:18:12 WebOS and other mobile OS
0:27:43 Flip cam killed by Cisco
0:35:30 back to Apple and Mobile Me
0:38:02 DECE wikipedia entry
0:42:49 Piracy
1:00:48 US Patent and Trademark Office
1:14:21 IVR discussion
1:26:15 Zimmerman, Crypto

IRC Transcript and, here are the index labels we spoke of, generated by Audacity (and me). The first column is seconds in to the recording after removing silences.

106    News of the World phone hackers
875     Foreign SIM cards, USA phone booths
1273    Is toll-free still relevant?
1581    Skype v5, and new linux version
1641    Security: Telepacific outage
1745   Security: attacked? Who you gonna call?
2250   Epsilon email db breach
2381    Comodo breach, bad DSL, bad
2631    Linphone, new versions including OS X and Windows
2775   g711, not that bad!
2887  iPhone SIP apps
3346   Lync USB phone
3957   Visio TV checking file system?
4964  Anyone using Google Talk VUC gateway?
5109    Index or transcript of VUC sessions
5442   IRC transcript of VUC

This month, many exclusive news items have been obtained by the VUC and we will be sharing these scoops with you this Friday.
photo: Scott Beale / Laughing Squid

Lots more discussion, 2 hours of it on various topics, continuing the chat about various platforms. Tablets, more Android, more on lag and latency (just guessing here, I went to dinner).

Dan York (insert typical remark about not him needing any introduction) jumped on the news that Google Voice was offering up SIP URI calling on March 7th, noting as his source a tweet from Aswath Rao (crediting @truvoip), all familiar contributors to the VUC. Then, Dan and many of us found it wasn’t working and he posted “Did Google Hang up…?” and then just two days later: “Google Voice Via SIP – It’s Dead, Jim”.

Plenty to talk about, but there’s a lot more news in Dan’s blog. Can he confirm that the AT&T/T-Mobile acquisition will make a new brand name in telephony: “T&A”?

Tune in and see!

Intro music by Zip Caplan.

Andy Kuan, CEO of Plum Voice is our guest. You may recall Plum Voice visited the VUC last year to show their Quickfuse app. He will be talking about a new product they offer, Floodlight, which addresses the need for a survey via web or phone. If you are into surveys, here’s a slide show about Floodlight. On that same page is a free trial demo you’ll want to check out. Key points he will be addressing are:

1. Overview of the Floodlight product and its origins
   • 17 different question types all of which work on both the web and the phone
   • Utilizes speech recognition extensively for a friendlier end-user experience
   • Integrated audio management
   • Integrated reporting
   • Runs on the Plum Voice infrastructure allowing calls to be terminated via SIP or TDM. With toll-free numbers in the US and Europe, or local numbers from much of the world via Voxbone.
2. Floodlight does more than surveys
   • Hooks for VoiceXML subdialogs for custom dialog handlers
   • Hooks in QuickFuse which can jump to surveys — use the right tool for the right job
   • Question-centric rather than flow-centric which makes it somewhat easier to understand for non-technical people
   • Data integration via SOAP

Another video conference solution to test: Visimeet from IOCOM.

This week, we’ll be talking about both phones and services experienced by VUC regulars and “drive-bys” alike.

If you have anything to report on phones, pbx, SIP trunking or any VoIP-related product or service, we’d like to hear about it.

OnSIP + Highrise

Karl give us the skinny on his new Palm Pre 2 phone and runs through the features of WebOS. They do sound great, but is there a futre? Will developers get on board or is this another OS/2?

Zac Wolfe and Eric Erickson join us from Safi Systems about news of their latest products.

SafiWorkshop is a call flow and IVR designer. With SafiWorkshop you can design, test, debug and deploy advanced call routing applications from a single, unified development environment. SafiWorkshop also includes a prompt recorder/importer for managing IVR prompts and a fully featured SQL query editor.

SafiServer is a standalone application server that supports a broad range of telephony applications and serves as the platform upon which everything from call routing to complex database driven IVRs can be built. SafiServer is built on enterprise level technology and has been designed from the ground-up with extensibility, scalability and reliability in mind. SafiServer can be run in Windows or Linux and has the flexibility to either be installed on the same system as your telephone switch (Asterisk or FreeSWITCH) or on a separate machine.

For those who adore SIP, open standards and open source, join us and meet AG Projects team, the makers of Blink and hear about the project, its inception and its future. Our guests are Adrian Georgescu, Dan Pascu and Saul Ibarra. These folks have done great things in the past, and we’re anxious to hear more about their latest work.

Adrian Georgescu, AG Projects founder and CEO, calls Blink “the new kid on the block” among SIP clients. Launched just one year ago, it managed to become for many in a short period of time their favorite SIP client on the Mac and now effort is underway to provide the same cool experience on Windows and Linux. Blink differentiates itself from its competitors by a generous feature set: g722 wide-band Audio from day one, File Transfer, Instant Messaging and Desktop Sharing, all those cool things SIP was all about, yet everyone stopped at VoIP. The lack of dial-pad also stirred some waters, it was the first soft phone to say goodbye to the classic dial-pad, a telephony oriented interface and offer instead a productivity oriented contacts driven interface. AG Projects has launched this month SylkServer, an open source product that complements all Blink features on the server side, and the two of them can provide now a Skype like multi-party conference experience by using pure SIP. With an impressive roadmap, proven capabilities and real customers behind, can AG Projects now make a difference in the crowded space of SIP and VoIP?

SylkServer allows creation and delivery of rich multimedia applications accessed by SIP User Agents. The server supports SIP signaling over TLS, TCP and UDP transports, RTP and MSRP media planes, has built in capabilities for creating ad-hoc SIP multimedia conferences with HD Audio, IM and File Transfer and can be easily extended with other applications by using Python language.

Much dust has been raised on VUC about all the ways to share video. This week, since our schedule was blurred by weather and other factors, the LifeSize conference has been put off until Feb 18th at the earliest. If you haven’t installed Lifesize yet, I recommend putting it off since the license, a $199 retail item, can be used in trial mode for 30 days only. Conferencing in the Hollywood Squares mode is one way to share video, but the broadcast model is something else entirely. Between now and then, I decided to revive the old Mogulus channel from the “Asterisk Users Conference” days. Mogulus became Livestream, and the product has matured considerably over the couple of years. You may recall our initial tests in 2007 with Mark Spencer. One of the interesting things about Livestream is that it is a very good presentation tool. If VUC guests, such as Voxeo’s Dan York have slides or desktops to show, this is an excellent way to go. Their presentation tool, Procaster (above, presenter and slides on one screen) is available free for Windows and OS X, as is the now ubiquitous Flash app that transmits the video and audio streams. The producer of the conference then can select who has center stage.  Livestream also has a backpack video production unit called the Livepack that uses 3G/EVDO to transmit from anywhere that has coverage for these services. (see diagram below: six 3G modems!) But I digress, as usual. First, I’d like to propose a toast. Then I propose a test of Livestream this Friday. No, it isn’t SIP and it isn’t open, but Livestream is a great way for our guests to present their products and services, and I hope some of them will do this soon. Sign up for a free account and send me your username by Friday so I can add you to the production team. You can choose to sign up to “Watch” which will allow you to participate in our VUC tests and future events or go all the way and sign up for a “Broadcast” channel. Both are free, ad supported and somewhat limited in reach, but it allows you to get your feet wet in the whole video channel production thing. That includes lower thirds, crawlers and other things like channel branding. Another interesting aspect of the facilities is recording. Ustream (and the other two similar video sites) does a lot of the same things, and may do some of them better than Livestream, but Livestream is more professional in my opinion. One of the reasons I was attracted to the Mogulus beta years ago was the feature called autopilot. When the channel is not broadcasting live, you can set the channel to show a storyboard of content you choose. You can also allow video on demand. In both cases you can also use their embeded players on any web site. The new channel, if approved will be http://livestream.com/voiptv. The problem with the older channel, which was approved years ago is that it is called “asterisk”. I hope to see that part straightened out before Friday as well. Stay tuned, film at 12.

The last minute cancellation of our Lifesize video conference experiment (postponed until Feb 18th we hope) left us with an open conference. Doug Mohney was kind enough to call in from ITEXPO, but because he was calling from the airport, the “connection” was lousy, so only some of the things he was trying to tell about got through.

Then there was a detailed discussion of the merits of Dollhouse as opposed to Tru Calling.

Thanks to @sollostech for bringing Bufferbloat to our attention!

Much more background is here: http://www.bufferbloat.net

What is Bufferbloat? From Jim Gettys’ explanation:

Bufferbloat is existence of excessively large (bloated) buffers into systems, particularly network communication systems.

Who is Jim Gettys? In his own words:

“As a quick synopsis, Bob Scheifler and I started the X Window System; I was the editor of the HTTP/1.1 proposed and draft standards, co-founder of handhelds.org and the revised X.org, and helped out on freedesktop.org.  I also worried a lot about the software for the OLPC XO-1. And a long time ago, in a galaxy far, far away, I used to astronomy.”

More details on Jim Gettys and his many projects and challenges on Wikipedia. We’re very pleased to have Jim as a guest and I know we’re anxious to hear more about Bufferbloat!

Dave Täht, who will also be joining us, has been involved in Linux and VOIP since 1993 and wireless networking since 1996. In addition to early work on asterisk and the
sofia sip stack, he worked on several early embedded asterisk devices. In 2006 he “retired” to Nicaragua where he worked on the OLPC, IP04, and mesh potato, in addition to surfing a lot.

After observing the (sad) state of the wireless networks and the internet itself, there, he :assumed: that the core problem was wireless
spectrum and started developing the wisp6 5.8 ghz wireless mesh network.

He was in the middle of the first test deployment this past spring and got beaten by both the arrival of the Survivor TV show… and bufferbloat (not realized until December of this year) , which forced him to return to the US. Upon arrival… He slowly realized that the
problems he’d had in Central America were also appearing in the US.

It wasn’t until December that he got a handle on it with Jim Getty’s series of postings on bufferbloat.

He is now spending all his spare time working on beating bufferbloat.

Randal L. Schwartz, [@merlyn] a name to conjure with…

Some of us met Randal at Astricon 2009 and he’s been a part of VUC whenever he’s had the time. Randal’s an author, a narrator, a programmer and a fun guy to hang with. Somewhere in the call about lag, packet loss and latency, we took advantage of Randal’s visit to catch up with what he’s been up to. One of his recent activities was doing some narration work for StarShipSofa, an award-winning Sci-Fi (oh, sorry, “speculative fiction”?) podcast. Randal has written and co-written several books on Perl: Randal Schwartz’s Perls of Wisdom
, Learning Perl (4th Edition) Covers Perl 5.8, Learning Perl Objects, References, and Modules.

Randal is also well-known as co-host on FLOSS Weekly the great open-source show, part of Leo Laporte’s TWIT Empire and for his long career in Karaoke.

iNum stands for international Number and is an initiative managed by Voxbone to develop an area code for IP Communications within the +883 global country code created by the ITU (International Telecommunication Union) started in 2008.  This will be an update to the earlier visit to the VUC made by iNum team members in January 2009.  Hugh Goldstein, Director of iNum Business Development will share current information and plans for 2011 and how VoIP users can leverage and support the iNum initiative in their own businesses and as individuals. Note that iNum supports voice, SMS and advanced multi-media services

Here’s Voxbone’s list of carriers who support iNum. (We notice that Orange Switzerland is not on the list, but we know that they can route iNum calls. They do this as surcharged numbers.)

Humbug Telecom Labs join us to talk about their latest release. Nir Simionovich mentioned this project when we last met (AMOOCON 2010 in Rostock? Seems like 100 years ago.)

Did you know that most modern PBX systems can be exploited? were you aware that your cellphone voicemail box can be exploited as well?

So begins Humbug’s about page. The following text is a slightly modified version of the bios of the three main dudes in the project:

Boaz Bechar

Backoffice and frontend developer. Being it mangling PHP code with CodeIgniter or snazzing up some jQuery magical interfaces, Boaz is the man behind it. Boaz is also the person behind most (if not all) of the analytical displays in the backend. Boaz comes to Humbug after spending almost 4 years with PokeTalk.Com, a Telephony 2.0 company – where he gained much experience in developing backends, billing systems, analytical methods and much experience with handling fraud and analyzing it.

Igor Ratgauzer

If you are seeking to learn about the various aspects of physics, math and programming languages, Igor is the person to talk to. With over 20 years of development experience, Igor brings to the team much of the development horse power required to undertake a task such as Humbug. Be it, PHP, PERL, C, C++, SQL or some other obscured language that may be required – Igor is the man for the job. Igor is the brains behind most of the mathematical engines required to make the HumBug platform work as efficient as it should. When you work the Humbug interface, you would notice that it is working extremely fast, in comparison to most other Asterisk related analytical tools – the core of that is Igor’s work and analytical SQL engines.

Nir Simionovich

Unlike Boaz and Igor, who spend most of their time writing code for the project, Nir spends his time dreaming about different things the project needs. Be it a new analytical engine that is required, or maybe some cool business prospect for Humbug, Nir is the person to talk to. Nir is a well known figure in the Open Source telecommunications industry. Being involved with the Asterisk Open Source PBX project since 2002, writing 2 books about Asterisk and public speakings at multiple expos & conventions, Nir is our telephony/telecom guru. Nir also maintains his own blog at http://www.simionovich.com and is also the CEO of a consulting company called GreenfieldTech.

Voxeo Director of Conversations Dan York joined us for a detailed explanation of what happened to the supernodes and what Skype engineers did to bring the network back up. Dan’s CNN moment is on Disruptive Telephony.

Skype has had a “monopoly” in this space for a long time. Also, the news just came out that Skype purchased Qik and Dan and a few of conjecture about what this means for the future.

In the second part of the call, after a brief excerpt from Silver Water, Leif Madsen, co-author of the Asterisk Book (aka TOFT) and Russell Bryant Digium’s Engineering Manager, Open Source Software tell us about the imminent release of the 3rd edition, “Asterisk, the Definitive Guide” and how you can help make it even better.

Superman Cartoon Theme by Zip Caplan. The “intermission” song “Silver Water” was written by Randy Resnick and Coleman Head.

Total Call Duration (not recording): 4h15m – Audio Legs: 245 (Max concurrent: 48)

TBA

A look back at 2010 in VoIP (4+ hours of VUC)

Here are a few things to get you thinking in advance:

• Apple’s Antenna-gate
• Apple’s Facetime over 3G
• Asterisk 1.8 Launches
• Amazon EC2 as a DOS platform
• Cisco launched UMI video conference device
• More HDVoice capable phones
• Skype Outage
• VoIP on mobile devices
• or perhaps something more personal…..

Come one, come all. Bring your story.

From e4 Strategies one HD-capable Gigaset DE380 IP R SIP phones (shipment to North America only)

VUC Holiday Special Edition

Today, our sponsor e4 Strategies is giving away two  HD-capable Gigaset DE380 IP R VoIP phones (shipment to North America only)

Polycom Productivity Suite licenses. We’ll be giving these away each week for the next few weeks. The suite adds the following features to your USB-equipped Polycom IP phone:

Corporate Directory Access
Local Call Recording
Visual Conference Management
Voice Quality Monitoring

From Voxeo Labs and Tropo

VoIP Users Conference (VUC) is probably the largest and longest-running weekly meeting of an international technology group. Callers join us live from at least 3 continents each week. This “Christmas Eve Edition” lasted 6 1/2 hours. The first two segments are just above. Here’s the final segment: 3 hours and 45 minutes. A few highlights:

Adam Kalsey of Tropo on connecting Skype to Tropo. I’ve tried just about everything but SMS and IM on Tropo for the moment. It’s hard to believe how easy it is to get started in a language of your choice: JavaScript, Ruby, PHP, Python or Groovy. This overview will convince you to get started, since it’s completely FREE to develop a Tropo app. Creating a simple IVR in PHP took me about 5 minutes, and because they have international DID available, I was able to see what it sounds like on the phone and Skype and even do proof of concept demos for my customers.

Michael Graves tries to get the Gigaset  DE380 firmware updated and connected in g722.

Dave Michels of Pin Drop Soup fame chimes in about Google’s Chrome OS netbook prototype

David Frankel of ZipDX talks about the REINVITE issues they face with non-standard SIP implementations

Crap support

Did I ever mention that support for Skype for Business is one of the worst I’ve ever experienced? Skype blocked one of my accounts last week and it took 4 days to get it unblocked, during which I told them “I can’t log in” and they wrote back with reasons my account might be blocked and suggesting I “log in to my account…” and I wrote back saying “but I can’t log in…” etc ad nauseum. Skype support sucks, no better way to say it.

Great support

On the other hand, Voxeo support, which I also experienced that same week, is excellent. I went to their #tropo IRC channel, and had answers within minutes and action on my account from a Voxeo support person. Kudos to Voxeo.

Cloud based telephony is becoming a ubiquitous new frontier of development for companies in this space. Recently, a blog post called “Why choose Tropo over Twilio” written by Tropo’s Chief Technology Evangelist (and Voxeo Co-Founder), Johnny Diggz, hit the front page of Hacker news over the weekend.

Comments on the new site erupted to a level just short of a flame war between the “super-angel” backed startup, Twilio, and Voxeo/Tropo.  Diggz will be joining VoIP Users Conference to talk about his role in this battle for the hearts and minds of a new wave of developers in cloud-based telephony.

More about Tropo

Twilio’s Evan Cooke responded to our (sorry, belated, my bad) invitation and talks about Twilio.

The Coalition to Salute America’s Heroes (CSAH) was created to provide a way for individuals, corporations and others to help our severely wounded and disabled Operation Enduring Freedom and Operation Iraqi Freedom veterans and their families rebuild their lives.

VUC is proud to support the 24-hour podcast event. The complete schedule is found on that link and if you like, you can volunteer to be there for a 30-minute segment to show your support.

During the event, the primary purpose is to raise money and awareness of the coalition, but we will be talking about new media, podcasting and the VUC of which you may be a part. I have asked to do a time that is morning for me, around 5 AM EST. If you are in a time zone where you are awake and active at that time, please consider joining us.

Todd Cochran explains his choice of  the 2010 charity in this post.

VUC, our international user group meets every week at 12 Noon EST. To see the time in your area, visit Permatime. This week, some of the regulars were called away but most of us did make cameo appearances. This week we spoke about mobiles and mobile data plans and access.

Our Google Group and Mailing List welcomes your membership. Be sure to give a brief intro, just a few words about your interest in technology.

Join us any Friday, we’re happy to answer questions and discuss tech topics of all kinds.

Linkedin: http://vuc.me/linkedin

Facebook: http://facebook.com/voipusers

Twitter: @VoipUsers

IPv6 is the next generation Internet Protocol (IP) address standard intended to supplement, and eventually replace, the IPv4 protocol most Internet services use today. To help ensure the continued rapid growth of the Internet as a platform for innovation, IPv6 tackles some of IPv4′s shortcomings – most notably a limited amount of remaining addresses. While the technical foundations of IPv6 are well established, significant work remains to deploy and begin using IPv6 capabilities.

Olle Johansson will be joining us to describe the state of the migration to IP v6 in VoIP-dom. Olle (@oej) needs no introduction. His company, Edvina was founded in 1987. Since then, they’ve promoted open standards, for networking, for application integration and for software. Olle is a pillar of our community and an outspoken critic of all that stands in the way of progress in his areas of expertise.

Proposed Issues on the discussion agenda
- IPv6 – how to get it into your network today
- VoIP and IPv6 – why is this a good marriage?
- Experiences from Asterisk 1.8 IPv6 support
- Living in a dual stack world
- Dual stack considered bad for VoIP?
- Implementations out there – a call for help? Which devices used in the VUC community supports IPv6? Any experiences?
- Setting up a VUC IPv6 testbed
- Follow-up call planning – monitoring the IPv6 migration process in the VUC community.

Olle is a born leader, but he wants YOU to stand up now and lead the rest of the world into IPv6!

“Get IPv6 up and running in your networks and learn how it works. It’s not rocket science but it’s neither exactly the same as IPv4 either. Let’s explore these unknown territories, let’s boldly go where no VUCer has gone before.”

Zeeek: “I’ve been a lot of places, but I’ve never been to IPv6!”

Some reference material:

Wikipedia on IPv6 – Pretty good article for the basic motivations and implementation info

IPv6 Act Now – News, interviews, forum…

The IPv6 Portal – organized by roles: press, user, developers…

IPv6 Forum – Deploying a cute IPv4 depletion counter

Article from the year 2000

Making room for the next billion users – IPv6 Internet Society

Guest: Jim Courtney.  Jim tells us about Skype-certified FREETALK Connect. Here are the technical specs.

By connecting your IP desktop and conferencing phones to your network, your phones will be detected and configuration files for your phones generated and pushed out to your phones.

• Auto-provisioning works with almost all models of desktop and conference IP phones
• Install wizard configures all basic networking, telephony system and user functionality on the FREETALK Connect
• Remote administration capabilities that enable the system to be administered from anywhere Internet access is available.

Jim Courtney writes on the web at Voice on the Web,  Voyces and other places…

Links for FREETALK Connect

Product Page/Data Sheet: http://www.freetalk.me/product/freetalk-connect/
Getting Started: http://www.freetalk.me/faq/freetalk-connect/
Reseller Application: http://www.freetalk.me/partnership/

Other links

Wireless Stereo Headset  with discussion of WiStereo: http://voiceontheweb.biz/2009/11/freetalk-freedom-wireless-stereo-headset-2-0-when-you-just-cant-sit-still-during-a-skype-call/

PamFax: http://www.pamfax.biz/en/
Pamela for Skype:  http://www.pamela.biz/en/
Calliflower: http://www.pamela.biz/en/

Phono, a new JavaScript library that allows you to build voice and chat/IM applications directly in a web browser, was announced October 16, 2010, at the JQuery Conference in Boston, by Voxeo Labs. What is phono?   A lot more info can be found here on the Phono Blog if you want to cut to the chase and see code snippets, and some inside views of what can do.  The Voxeo team unveiled Phono just a few weeks ago. Phono is an open-source SDK to embed telephones and instant messaging services in your web pages. With Phono, you can (among other things) add click-to-call buttons to your call center or your applications hosted on one of Voxeo’s platforms (and you will eventually be able to host your own server-side infrastructure soon).

Download phono

We added the WordPress Plugin at the right in 30 seconds! It allows your visitors to make phones calls right from your pages. A headset is recommended if you want to actually call in to the VUC during our Friday conference with Chris Matthieu. If you run WordPress, go to Plugins, enter “Wordphone” in the search and install away!

Links mentioned on IRC

SIP Communicator – the Java VoIP and Instant Messaging client.

SIP Communicator is an audio/video Internet phone and instant messenger that supports some of the most popular VoIP and instant messaging protocols such as SIP, Jabber, AIM/ICQ, MSN, Yahoo! Messenger, Bonjour,  and a whole lot of other useful features. In the latest builds, SIP Communicator has g722 capability, making it a choice for use with VUC via ZipDX.

SIP Communicator is completely Open Source / Free Software, and is freely available under the terms of the GNU Lesser General Public License.

SIP Communicator has binaries built for Mac OS X, Windows and several linux distros, and the source is available on the SIP Communicator downloads page.

Asterisk SCF is a framework that will allow developers to move beyond voice and also create open source video and text that can be used for anything from embedded applications to enterprise and carrier solutions, according to Digium. It is delivered as a system of distributed components that can be deployed in clusters on a single system or on many systems, and will support pretty much all real-time IP communications including video, wideband audio, chat and desktop sharing.Asterisk SCF seeks to meet four requirements for businesses looking to deploy open source communications solutions:

• Performance
• Scalability
• Fault tolerance
• Extensibility

Live from Astricon (well it was when we recorded it), Kevin Fleming, director of software technologies at Digium joins us at talk about the big Digium announcement.  This discussion was made possible  by Tim Panton and Michael Graves. Kevin, Tim and Joshua Colp were in a room, surrounded by John Todd’s tie-dyed shirts.

Double innovation, two topics this week as Counterpath has two big stories. Todd Carothers, VP Product Management and Mike Doyle, VP Technology will be on board to tell us more about these two developments and to answer your questions.

Counterpath was granted a patent (# 7,809,381,) for presence detection in mobile and fixed broadband networks.

By providing applications with a graceful way to automatically select the best network connection based on the user’s location, the patented mechanism enables services such as least-cost routing and minimizes dropped calls by streamlining the process of transferring calls from the cellular macro network to Wi-Fi or a femtocell.

The new patent announced today leverages these two patents to enable the delivery of the voice and data services based on the end user’s location. For example, when a person moves from a mobile network into an IP-based network – such as one in a home or office – the combination of these three patented mechanisms allows the home or office phone to automatically assume the user’s mobile identity, or mobile phone ID. When the user leaves the home or office environment, the phone is seamlessly disconnected from the mobile identity and moved back to the mobile network.

Counterpath Bria for Android

Compatible with any smartphone running Android V2.1 or later, Bria Android Edition’s features include:

    -   Support on any SIP-compliant server to enable rapid implementation
    -   Multi-tasking support for background operation, such as fielding
        incoming calls while using other applications
    -   Automatic codec selection to ensure optimal call quality
    -   Deskphone-class options such as rejecting, holding, forwarding,
        merging and splitting calls, as well as attended and unattended
        transfers
    -   A detailed call history pane that displays dialed, answered and
        missed calls, along with ability to delete entries
    -   Advanced security settings, including audio encryption
    -   NAT traversal to balance security and ease-of-use
    -   The ability to work with the native dialer
    -   Integration of user contacts stored on the device
    -   Voicemail
    -   Optional customized branding for graphic assets and SIP settings
        available for enterprises and telephony providers.

Many VUC members are testing the beta now and they will be able to interface with the Counterpath folks during this call.

and boy, are my arms tired! But wait! It gets better!

Part 2 may be an acquired taste, but some people really like to listen to these, so here it is.

My advice? Be there. Join us every Friday at 12 Noon Eastern Times, standard or daylight, which ever is in force that day. There is a permatime link to know when the next session is in your very own time zone.

On the vuc.me web page during the live conference there is a link to the new experimental streaming URL and a widget that allows you to call VUC from a browser. We’re in IRC channel #vuc on Freenode.net.

If you’re reading this (and I know you are!) make a note to call sip:200901@login.zipdx.com or skype:vuc.me next week. Why? Because we don’t record the last 5 hours of the VUC. We might be talking about you.

[6:23pm] Alok_: correction – http://truvoipbuzz.com
[6:27pm] mhill: http://toddcarothers.com/2010/10/caught-at-itexpo-sms-for-itsps-demo/
[6:28pm] maximCH: http://dilbert.com/strips/comic/2010-09-27/
[6:31pm] DelphiWorld:  http://www.inum.com and other VoIp and ITSP providers to ofer sms over sip
[6:38pm] jtodd: Sounds like this patent: http://bit.ly/a6chDc
[6:39pm] mjgraves: UMA = http://en.wikipedia.org/wiki/Unlicensed_Mobile_Access
[7:16pm] JDBalogh: I wonder how it compares with http://divitas.com/ that was doing this 4 years ago?
[7:21pm] JDBalogh: http://forums.counterpath.com/
[7:23pm] Zeeek: https://newborn.malaiwah.com/stream/vuc-me.m3u
[7:30pm] dhartman: http://www.youtube.com/watch?v=a3C_wLCZ18M
[7:40pm] televoip: http://toktumi.com/
[7:40pm] maximCH: http://www.toktumi.com/
[8:24pm] Zeeek: http://vuc.li/vucstream
[8:42pm] JDBalogh: kfife: http://www.voipsupply.com/wappoe12
[8:43pm] JDBalogh: If the supply is 5V and you have 802.3af, use this: http://www.8774e4voip.com/Linksys_POES5_p/linksys%20poes5.htm
[8:47pm] JDBalogh: kfife: 5V at 3A according to http://www.nortel.com/products/01/succession/es/collateral/nn123291.pdf
[9:10pm] dhartman: http://mirror.astlinux.org/downloads/iso/geni586/astlinux-0.7-4568-asterisk-1.8.0.iso <– if you want a live ISO for testing Asterisk 1.8.0
[10:29pm] p3nguin: zeeek: If you know who the person is, you can always send them something from this site: http://www.poogram.com/
[11:20pm] leifmadsen: http://ofps.oreilly.com
[11:28pm] ILNetGuy: http://en.wikipedia.org/wiki/Datagram_Congestion_Control_Protocol
[11:46pm] GregW__: Amazon EC3 Free Tier for one year http://aws.amazon.com/free/?tag=gmgamzn-20
[11:49pm] Zeeek: http://www.spinics.net/lists/asterisk/msg133769.html

Please compare you wget or curl times for the following two instances of the Part 3 file and post results as a comment here:

http://c1406282.cdn.cloudfiles.rackspacecloud.com/280bis.mp3

http://cloud.voipusersconference.org/280bis.mp3

Do people ask you to work for them for nothing? I just had a long exchange of emails today about this ending with “thanks for your generous offer to exchange one hour of my professional time for one hour of yours. I want to learn, so I guess I’ll just muddle through it.” However, “learn” seemed to mean “ask Randy how to do it and continue to send questions and screen captures by email”. I offered to fix her problem with DNS in a few minutes by just going in to the interface and setting the CNAME and MX the way it needed to be done.

Also, the new 855 toll-free numbers and how to reserve one. From $30-99, get your name in a new number. Is there any public awareness of these?  More on Resporg and how numbers are moved around. I have tried to reserve on on Vitelity, we’ll see how that goes.

A small sample of topics covered:

FCC ruling about warning customers when they are going over their limits to avoid “bill shock”.

Mobile technologies, what are they about? GSM, GPRS, UTMS, HSPDA, CDMA, 3g, 4g?

Challenges of developing for OS X and why betas begin life for Windows only (such as Skype 5).

Remember pagers (beepers)? Remember heavy brick cell phones.

Wimax technology.

Wikipedia: As a standard intended to satisfy needs of next-generation data networks (4G), WiMAX is distinguished by its dynamic burst algorithm modulation adaptive to the physical environment the RF signal travels through. Modulation is chosen to be more spectrally efficient (more bits per OFDM/SOFDMA symbol). That is, when the bursts have a high signal strength and a high carrier to noise plus interference ratio (CINR), they can be more easily decoded using digital signal processing (DSP). In contrast, operating in less favorable environments for RF communication, the system automatically steps down to a more robust mode (burst profile) which means fewer bits per OFDM/SOFDMA symbol; with the advantage that power per bit is higher and therefore simpler accurate signal processing can be performed.

I’m more confused than before.

VUC is an international group that meets every Friday at 12 Noon Eastern Time. We discuss technology and often VoIP, telephony, the Internet, web sites and hosting, networks and networking. VUC is @voipusers on Twitter. This week, no specific guest product or service made the discussion wide-ranging, with participants from Hong Kong, Algeria, France, Switzerland, Austria, Holland, Spain, Barbados and even the USA. A few of the subjects covered this week:

Cisco umi, is this worth following? Is the concept of video conference on your TV even something there’s an audience for? Do many people have the needed bandwidth for 720p or 1080p?

Counterpath launches a beta trial for there Bria Android app.

Who still have landlines? Why?

TV standards like PAL, NTSC, Secam…

David Duffet on the Ringer Rodeo at Astricon.

AG Projects’ Blink was bridging the ZipDX wideband conference bridge and Talkshoe for 3.5 hours without a single glitch.

Join us every Friday! If you have a product or service you think would be of interest, please reach out to us for a guest spot using the contact form or by grabbing @VoipUsers on Twitter. Join our mailing list or read it on the web as a Google group.

Thanks to those who requested this on our mailing list! We have liftoff! (VoIP Security)

VoIP Abuse Project . J. Oquendo

“Arkeos is a program primarily used on Unix based servers running the Asterisk Open Source PBX. The application mimics a valid extension in which an attacker, after bruteforcing an account, attempts to place calls. What the attacker doesn’t know is that their calls go nowhere. The sole purpose of allowing them to perceive they have an account is to track them. Where they come from, what accounts they use, what VoIP account scanners they use, what numbers they try to dial. Because of the flexibility of Asterisk and Unix as a whole, the application does a lot more than stated however, I will not disclose too much otherwise any attackers reading this might catch on. Then I won’t have anyone to play with.”

From Jason Goecke’s blog post – read the whole post here on the Tropo blog. Jason made the announcement to the world about this new project on the VoIP Users Conference!

“We have decided to take it a step further and created a Tropo Scripting application that turns Tropo into a giant Asterisk application platform in the cloud.   You can now run just about any Asterisk AGI application on Tropo.

Tropo AGItate was started on the Nerd Bird (good to have in-flight WiFi) from San Jose to Austin, on my way to LoneStar Ruby Conference. Jim Freeze – the organizer of LSRC – had recently been to AdhearsionConf in San Francisco; I wanted to be able to show something extra special during my talk there. On that one flight, I was able to get the basics working and show Tropo emitting AGI during my talk, just like that. (For those non-Asterisk folks out there, AGI is an API that lets external applications connect in to Asterisk and fully control it).”

Key points:

• Tropo can now speak AGI over TCP to any FastAGI server, including Adhearsion, PHPAGI, Asterisk-Java, etc.
• A long list of Asterisk commands are supported, and of course all the Tropo ones.
• While the script is written in Ruby, no Ruby knowledge is necessary to use it. Just point your Tropo application to the Ruby script on Github, upload a configuration file via FTP or WebDAV to your Tropo account, and you are ready.
• Full support of Tropo Speech-Synthesis (TTS) and Speech-Recognition (ASR), but also for Asterisk sound files. Yes, tt-monkeys works!
• Fail over to a SIP URI, in case your FastAGI server does not respond. This could be to another Tropo application, another Asterisk box, or anything that supports SIP.
• You get all of the Tropo channels over AGI, including SMS, Instant Messaging and Twitter, all using the same application.
• Support for custom SIP Headers in and out of Tropo.

Tony Lewis, Founder/Owner of Schmoozecom stopped by to tell us about their product, Pbxact. They say, “Say NO to feature codes and hard to remember sequences of key presses! Phone applications let you intuitively control your phone calls and are included on every phone and every PBXact system. Our easy-to-use applications give you: visual voicemail, do-not-disturb, parked calls, intercom, groups, transfer-to-voicemail, call forward, day/night mode and away statuses.”

There’s no way to do this without spenging the entire morning on it, so here without further apology, are the links pasted into the IRC channel #vuc on Freenode.net. THese links are related to all three subjects for Friday October 1st, 2010.

Voipsa, Voipsec, J. Oquendo’s work on VoIP Abuse Project

http://voipsa.org/blog/2010/09/28/voip-abuse-project/
http://voipsa.org/blog/author/joquendo/
http://www.linkedin.com/in/voipsec

http://www.securityfocus.com/tools/45

http://www.honeypots.net/honeypots/products

http://www.gnuma.org/detail/351/deception-toolkit.html

Randall Schwartz: snort has asterisk rules http://www.snort.org/search/results?q=asterisk&action=Search
artemisa is the other honeypot i was speaking of http://artemisa.sourceforge.net/
SIP Bastion host paper http://www.snocer.org/Paper/snocer_D2_2.pdf

http://www.projecthoneypot.org/list_of_ips.php

http://www.infiltrated.net/voipabuse/addresses.txt

http://www.infiltrated.net/voipabuse/subforms/

http://www.ingate.com/Products_siparators.php

http://www.voipplanet.com/backgrounders/article.php/11924_3775451_2

Tropo AGItate

http://tropo.com/agitate/
@Tropo can now speak AGI over TCP to any FastAGI server, including Adhearsion, PHPAGI, Asterisk-Java, etc! – http://blog.tropo.com/
@Tropo AGItate is open source and now on @GitHub -

https://www.tropo.com/how-it-works/

https://www.tropo.com/pricing/

http://www.voip-info.org/tiki-index.php?page=Asterisk+perl+agi

http://www.voip-info.org/wiki/view/Asterisk+FastAGI

http://search.cpan.org/perldoc?Asterisk%3A%3AFastAGI

http://www.vrplumber.com/programming/starpy/

http://hackage.haskell.org/cgi-bin/hackage-scripts/package/AGI

http://tools.assembla.com/erlast

@randal http://infiltrated.net/voipabuse/addresses.xml gonna whip this up during the course of the day and script it to keep it updated

Schmoozecom and Pbxact

Schmoozecom is giving away an Aastra 6739i this afternoon.
Calling the VoIP Users Conference with PBXact  — http://www.youtube.com/watch?v=jByzQvnzOrE

http://videos.pbxact.com/Phones/

http://www.aastratelecom.com/cps/rde/xchg/03/hs.xsl/23485.htm

Going back.. here is the screen while programming http://images.pbxact.com/PBXactHardware/UnBoxingClassic/aastra3.jpg

http://www.sip-communicator.org/

http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation

Like a heart-broken lover, after Wave, I hesitate to get excited about the open source projects I hear about. Wave never really panned out, and personally I think there was too much T & A and not enough brains and horsepower there. All show and little tell. On the other hand, there’s ThinkUp, a project just emerging from the ashes of its previous incarnations and looking refreshed and energetic.

ThinkUp (Expert Labs Project) is a PHP/mySQL program that will scour social media sites for its members and turn the cacophony (some would say CACAphony) it finds there into a nearly coherent conversion. Currently working with Twitter, Facebook and Flickr, it has a sporty plugin architecture and a simple, modern, clean look to its pages.

Wonderful, you say, drifting off, but what does it do?

I kinda wondered about that myself, so instead of setting up the crawler as a cron job, I ran it manually. Anyone into the open source world will enjoy watching the crawler cheerfully tell you what it is doing, babbling away with just enough info about how hard it’s working and how much juice Twitter is allowing you to suck in until next time. The crawler looks at the accounts you’ve set up, grabs the posts, Tweets, for example and stores them in the database. The data is sliced, diced and messed with and displayed in many different useful ways. I guess if you wondered about the expression “wisdom of the crowd” which I’d challenge daily, here’s a tool to test that very idea. One of the plugins available will decode short links, another does something with geolocation that is probably worth doing.

Allow members of your crowd to sign up for your ThinkUp site. They then connect your ThinkUp app to Twitter (and the other sites if you care to). Assuming people are talking on Twitter, and that they have allowed their posts to appear on the public timeline, you should be able to see conversations, whole subjects developing and interlinked.

Head over to Gina Trapani’s Github and grab the code. It literally took me 5 minutes to install and get it running. ThinkUp requires a server with php5,  mysql and a minimum of pep if you plan on having members and guests. Give it a try or wait until we open our own VUC ThinkUp.

On September 16th, 201, an Internet outage that had wide-ranging effects of disruption happened and Karl Fife wants to know who to punish! Interesting thoughts and observations. John Todd educated us on DNS resolvers and such. Very good if geeky discussion, our warmup to an exceptional late night (for me) session with David Rowe.

If you haven’t been made aware of Voyces yet, they define themselves as “Seven industry professionals providing insight, access and information from decision makers, for decision makers”. We define them as “required reading”. It is our hope to have each of these  industry “voyces” come and chat with us on VUC.

This week’s guest is Larry Lisser (@LarryLisser), who co-founded StartupCamp Telephony, an event focused on exposing innovation happening in early stage communications companies. StartupCamp2 will be held in Los Angeles on October 4th, 2010. Last year’s event in Miami was standing-room only.

Note: We often joke about the bad quality of callers from the professional world of VoIP and this week was my turn (Zeeek @VoipUsers) as I had my mixer connected wrong. As a result there is echo during part of the call. My bad, sorry!

“High Definition Voice” on phone conversations is something you may not notice until you have experienced either very bad or very good quality. Andy Abramson of VoIP Watch will be joining our discussion of HD rollouts by [a carrier whose name begins with O] and the ramifications and challenges as he expressed them in his recent article.

Actually, it was before hours, but who’s counting? Talk about Apple TV, Cisco wanting to buy Skype, Skype for Business (was SIP for Skype), bluetooth headset woes and much more.

SIP for Skype is now called “Skype Connect”. We’re wondering about the cost of the service. At one point I heard there would be a base charge for Skype Manager and then a charge for each user, plus a per channel charge. If this is the case, I can only say “Get a Life, Skype!”

Some intense examination of the requirement to tell people you are recording a call with the regulars while waiting for FS bridge to successfully UNmute it self. Note that ZipDX implemented the warning when you connect to the bridge if recording is on. Some reminders about *6 to toggle mute state.

The origin of “Look to your skies for a warning! Keep watching the skies!” Naw, I cut that part but I wanted to add the link in this post, because I didn’t get more info from FS about what to put here. Great movie, “The Thing from Another World”, featuring James Arness (“Gunsmoke”) as “The Thing”. One of the best science fiction pictures from the fifties, and one that helped define the genre.

From yet another world, (the Freeswitch bridge), Pacman sounds, and then belched forth the actual meat-space aural personna of the like of Brian West, Michael Collins, Anthony Minessale, Darren Schreiber and a few less-stellar voices…

From their site at Freeswitch.org: “Created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many telephony applications can be developed using a wide range of free tools.

FreeSWITCH was originally designed and implemented by Anthony Minessale with the help of Brian West and Michael Jerris. All 3 are former developers of the popular Asterisk open source PBX.  The project was initiated to focus on several design goals including modularity, cross-platform support, scalability and stability. Today, many more developers and users contribute to the project on a daily basis.”

The final part of the Freeswitch sessions. The ever-efficient Karl Fife wrings yet more tidbits of useful information from whoever was left on the Freeswitch bridge. This would have been in the fourth or fifth hour.

Freeswitch book is here. The official title of the book is FreeSWITCH 1.0.6, however don’t let the number fool you. The information applies to both FreeSWITCH 1.0 and 1.2 branches.

Dave Michels‘ son Dustin plays a mean guitar. Here’s  Dustin Michels video.

With everything up in the air, we weren’t sure they’d be a session, but I arrived back in time, so I should be there to talk about whatever anyone brings up.

By a bit of good luck, Russell Bryant stopped by and shared a lot of interesting details and answered question about what’s new with Asterisk. Russell is Engineering Manager for the Open Source Software team.

We later got into talking about the 3G used, unlocked and jailbroken iPhone I bought in the USA and broght back to Europe without knowing how it would work. I was saying that in California, the iPhone showed my location to withjin feet without data or wifi connection, as long as the map was already loaded (using wifi at home). That brought us to GPS and how it works. Here in France, the location it finds is much less accurate.

Extended discussion of GPS and how it works. Triangulation or strangulation?

Freetalk Connect wants to put Skype on every phone in your office.
Designed for offices with between two and 50 users, the FREETALK Connect offers  unified communications functionality, including Find Me, Follow Me; a unified voice mailbox; automated attendant and auto call distribution.

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.

Allison Smith (@VoiceGal) pinged me about the collection she recorded for Joey Lindstrom for use with Asterisk pbx. The full set, which Mr. Lindstrom has generously decided to place in the public domain is available for dowload free here:
http://vuc.li/FunnyAllison.

Joey wrote to say this was his way to give something back to the community, and he’s asking Digium to put these prompts in the Asterisk distribution should they wish to do so.

2600hz is home to a collection of open-source telephony software that enables the use of the FreeSWITCH, Asterisk and YATE switching libraries. Initially built around the blue.box project, we aim to provide a collection of software to power your GUI, your cloud-based telephony switch and/or your monitoring and maintenance tool set.

Video testing – Next week we’ll be doing a real video conference, watch the site for more info

ClueCon - No one wished to comment. Whatever.

OpenVBX is a web-based open source phone system for business.
With a Twilio account and a web server with PHP 5.2+ and MySQL 5, you can build your own “hosted pbx”.

Build your own custom phone applets with just a little bit of PHP. Rebrand and resell OpenVBX to your customers.

Give every user their own phone number and personal conference line. Dial whole departments, share voicemail messages with the team. OpenVBX is for companies and collaboration.

Adam Ballai, lead engineer on the OpenVBX project and Twilio CEO Jeff Lawson are our guests.

Some hard info to be had about the do not call list and a lot of messing around with microphones, AGC and iPhone SIP clients.

Everyone joining the VUC this day will get a free ticket to AdhearsionConf!

AdhearsionConf has been announced and registration is open. Jay Phillips, the creator of Adhearsion, will be present for two days of talks, discussions, hacking and pair programming on all things Adhearsion. Other folks will be  there sharing their innovative uses of Adhearsion.

John Higgins, a Developer Outreach guy at Voxeo Labs joins VUC to um.. reach out to developers and get the Adhearsion word out.

Adhearsion is a new way to write voice-enabled applications. It’s not just an API or library — it’s a fully-featured framework, the first of its kind, designed for maximal code reuse and intuitiveness. The name “Adhearsion” is a combination of “adhesion” and “hear” because Adhearsion shines best when integrating technologies with voice.

Adhearsion uses the high-level Ruby programming language and has a comprehensive regression test suite. In the age of the social web, what other technologies are more social than voice? Adhearsion aims to help merge this so-called “Web 2.0″ world with the previously untamable telephony world.

It’s intuitive

Voice applications don’t have to be hard. Adhearsion abstracts much of the complicated telephony domain, allowing you to focus on what’s really important: the business domain. Take a look at the Examples to see what we mean.

It’s open-source

We take open-source software seriously. The telecom industry, above all, needs fresh innovative talent and Adhearsion’s here to foster it. This is one of the final major frontiers for open-source to truly make a transformative impact and now is an exciting time to be a part of it.

Guests: Steven Johnson, President and Product Support Director Scott Beer of Ingate Systems AB, a Stockholm, Sweden based high-tech company that designs, develops, manufactures and markets data communications products for trusted Unified Communications. Ingate designed the world’s first Session Initiation Protocol (SIP)-capable firewalls and SIParators, products that enable Unified Communications over the Internet.

The Ingate SIParator® is a device that connects to an existing firewall to seamlessly allow the traversal of SIP-based communications. Ingate SIParators are compatible with all existing firewalls and operating systems.

The Changing Role of Session Border Controllers:  Ingate’s SIP-capable E-SBCs protect and enable SIP networks worldwide.  E-SBCs used to just serve a primary role of security.

We will be discussing the following pooints:

- Talk about the security role E-SBCs play.  One question many people ask us is, “Why have an E-SBC in a SIP trunk deployment if the service provider is taking on security?”  There is an excellent answer to this, one that every end-user needs to know.

- Address common security mistakes we see in SIP implementations.

- Discuss how E-SBCs enable SIP trunking and other SIP-based applications, like video, citing case studies

- Talk about Unified Communications.

Quickfuse offers a graphic interface which allows you to drag and drop your way to a full-featured IVR application and have a working prototype in minutes.

Quickfuse incorporates access to data, logic and branching, instant visual feedback and a free trial account.

Integrics is the creator of Enswitch, a complete highly scalable and redundant integrated solution for commercial telephony services such as hosted PBX, ITSP, SIP trunking, and calling cards. It’s also the creator of Geons, a new project for distributed enterprise applications such as distributed PBX. Integrics CEO Alistair Cunningham will be talking about these products, building highly scalable systems using Asterisk and OpenSIPS, and answering your questions.

Geons 0.4.0.
This is the first version for public release. It comprises geond (the server), the web interface, and the basis of a distributed PBX system. For more details, including links to download and installation instructions, please see this page.

Geons is a suite of software for distributed enterprise applications. Geons applications:

• Are highly scalable, supporting millions of users.
• Can be geographically distributed across thousands of sites.
• Are fault tolerant, including geographical redundancy.
• Support disconnected operation, where a site is temporarily disconnected from the rest of the network.
• Are highly customizable. All templates can be edited and properties added, changed, or removed.
• Are extensible. New templates can be added to create new applications and enhance existing ones.
• Can be integrated with other software using an open XML based protocol.
• Are secure, with full flexible access control lists (ACLs) for all objects.

Both SIPVicious and Adhearsion have announcements to make. Sandro joins us to explain “svcrash”, a new addition to the tool kit to stop attacks when they happen:

A new tool has been added to SIPVicious – svcrash.py. As the name implies, it crashes something – svwar.py and svcrack.py. This tool is meant to be used by system administrators and organizations that are receiving unauthorized scans on their exposed IP PBX.

Ben joins us to give some details about the evolution of Adhearsion which has had a recent version release:

This release marks the continuation of the platform with some major enhancements and bugfixes:

• Support for Asterisk 1.6 (continued backward compatibility with Asterisk 1.4)
• Added support for ActiveLDAP
• ActiveRecord now works properly with Adhearsion components
• Daemonizing of Adhearsion no longer truncates log files
• Escape commands sent to Asterisk via AGI
• Asterisk Manager Interface (AMI) events now work when daemonized
• Various enhancements for improved Asterisk 1.6 support
• Various bugfixes