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.

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

OnSIP + Highrise

[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.

Talkshoe: [audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-224822.mp3|titles=Skype For Asterisk Tests]
ZipDX/CloudFront: [audio:http://cloud.declic.com/Part1Cloud.mp3|titles=Cloudfront verion of ZipDX recording]

Today we talked about the EXPERIMENTAL link using Skype for Asterisk (SfA for short). The more we use SfA, the more the Skype people will see that we the Asterisk community are serious about it.

The whole thing began as a phone call made by Tim P. to my Skype number. I almost never have Skype on but the number is forwarded via Gizmo5 to my phone, so when I heard it ring and saw the Skype ID, I realized the call was going through a bunch of servers and trancodings. The quality of that call was striking: total silence when no one was speaking (the equivalent of very black on an LCD display) and very good quality audio with a comfortably small latency. The call lasted at least 30 minutes, too and remaind of good quality.

So, we had at least 6 people called in via Skype from the USA, Israel, Barbados and the UK. The quality varied with the number of channels in use and time, but it was pretty good by and large.In fact, the quality when there was no packets dropped sounded good, even though it was g711.

IRC Text Transcript 2009-06-12

[audio:http://media.blubrry.com/winelover/recordings.talkshoe.com/TC-22622/TS-81881.mp3]

Junction Networks CEO Mick Oeth will present OnSIP, a “completely customizable sip network for hosted pbx, virtual pbx or any business communication need. OnSIP takes SIP to the next level, with free domain hosting, support for any SIP device, no user fees or seat licenses”.