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

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.

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!

OnSIP + Highrise

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.

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.

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.”

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!

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.”

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.

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.

Martin Sunstrum, the founder of Aksys Networks joins us for a presentation of the KONNECT line of business VoIP phones.

He will give an overview of the KONNECT Business Phones and explore the tight feature integration with Asterisk-based PBX systems.

Lastly, he will present the intriguing KONNECT Office Phone System, a multi-site office phone system that requires no PBX or hosted service.

[audio:http://recordings.talkshoe.com/TC-22622/TS-333211.mp3|titles=pbx Choices]

A fight to the finish with VUC’s resident VoIP skeptic Dave Michels and Mike Oeth, CEO of  Junction Networks (OnSIP.com) on choosing between hosted and on-premises pbx.

What are the arguments for and against a hosted pbx for your business. Are they different for small and larger business pbx choices?

Dave and Mike fight it out with the help of referee Michael Graves and the roar of the crowd.

VoIP online community and VoIP how-to

Voipathon will be talking about online community, dealing with censorship, countries that intentionally block telephony to prevent communications and much more. Using voip, connect with your online community and accomplish great things (or just socialize, we do both!)

Join us for Voipathon Live!

Honolulu: Thursday 25th at 9 PM
San Francisco: Midnight Thursday night to Midnight Friday
New York: 3:00 AM Friday
France,Germany: 8 AM Friday
Mumbai: 12:30 PM Friday
Beijing, Singapore: 3 PM Friday
Sydney: 6 PM

VoIPathon will be held for 24 hours, beginning on March 26th at 3 AM Eastern Time. Subjects discussed: online communities doing good things via the Internet. Online community helps you find like-minded people to learn from, share info and get things done.

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-222071.mp3]

Part 2: Yate and g722 [audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-227343.mp3]

YATE with Diana Cionoiu, who always shoots from the hip. Yate is an open source software developed with reliability, scalability and flexibility in mind for tomorrow’s telephony networks. I guarantee an interesting call this week, but then we always have interesting calls, right? Otherwise, why would they last up to four hours like today’s call?Here is a link to Diana’s presentation at Asterisk Tag 2008. She was also at AMOOCON 2009, here’s the page for her presentation. (Almost all the presentations are now available on the AMOOCON site. Nice work, Stefan!)

Acting upon the suggestion of our group, here is the IRC transcript of the call which again lasted over 4 hours.

IRC Transcript 2009-05-22

Part 2: Yate and g722

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-190012.mp3]

IRC Transcript 2009-02-20

Jason Fischl from Counterpath joins us to talk about their SIP clients and more.

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-184932.mp3]

Allison Smith, the Voice of Asterisk

Allison Smith, aka “the voice of asterisk” is always a fun guest to talk to but her contributions to the Asterisk community are ongoing. We talked among other things about her new site, The Asterisk Voice.

Membership on TheAsteriskVoice.com is growing steadily, and is proving to be the ultimate clearing house for Asterisk stock prompts. Take those stock prompts a step further, and enlist Allison’s help in customizing and personalizing your company-specific messages.