mike's blog

We are continuing to track down an issue with calls that are forwarded to Junction Networks DIDs (phone numbers) from outside carriers. We have escalated the issue with our carrier and will continue to update with the progress. Outbound calls are unaffected. Direct inbound calls to Junction Network's DIDs are unaffected.

We are currently tracking down an issue with calls that are forwarded to Junction Networks DIDs (phone numbers) from outside carriers. We have escalated the issue with our carrier and will continue to update with the progress. Outbound calls are unaffected. Direct inbound calls to Junction Network's DIDs are unaffected.

It has finally happened and I couldn't be happier. When I first got my iPhone I couldn't wait for the time when I could make extension to extension VOIP calls via my iPhone. The iPhone has wifi access. All it needed was someone to put some SIP software on the phone and, combined with Junction Networks OnSIP hosted PBX, it would be ready to go.

That time has come. Over the weekend I downloaded fring and after just a little tweaking I got it to work. The device registered to Junction Networks and my first call was to extension 7008: Tim in Chicago. All I had to do was bring up the 'dialer' in fring and dial 7008 and hit the 'SIP' button and it dialed Tim's extension. Tim only saw a call from me, ext. 7001. As far as he knew I was at my desk. In reality I was 30 feet away on my iPhone speaking to him over the iPhone's wifi connection. Next, I told Tim to hang up and call me right back. He dialed extension 7001 and fring notified me on my iPhone that I had an inbound call.

The calls sounded great. There was no echo and no delay. No question the call was better than a normal cell-phone phone call. I shared my story with fring and received the following response from fring co-founder and Chief Technology Architect Boaz Zilberman “fring prides itself on being a best-of-breed mobile Internet communications and community application, offering its users the ability to talk, chat and interact with fellow fringsters and all their online buddies using their mobile Internet connection. Partnerships with forward-thinking companies such as Junction Networks offer a new level of openness and connectivity and provide a glimpse of the future in which mobile Internet and traditional telecoms converge to provide as rich and flexible a user experience as possible.”

The best part is that extension to extension calls are FREE. Junction Networks never charges for extension to extension calls, and since fring calls use the wifi network and not the cellular network, it does not take up cellular minutes.

The fring dial pad interface does not have buttons for actions like 'Hold', 'Transfer' and 'Conference' so those features do not work in the fring interface. DTMF (touch tone) is disabled during a call. It would also be useful to be able to put SIP addresses into contacts and dial a contact's SIP address as opposed to being limited to dialing only numbers. Even with those restrictions, fring is a great start.

AT&T just announced that they are offering free wifi to all iPhone subscribers at any AT&T wifi hotspot. This includes Starbucks. Customers can locate Wi-Fi spots through AT&T's online tool or can locate a Starbucks using this tool.

With Junction Networks, since we allow our users to be registered on up to 10 different devices, this is a free addition to our service offering. There are no additional charges to add a device to the system. (Nor do we charge per user, so you could add a new user and setup the fring device as a user separate from your normal user as well.) All of our users can now setup their iPhone (or other mobile device) to act as their extension on their company PBX. No more 'forwarding calls to your cell phone'. Now, your cell phone IS your extension!

Press/Blogs
Andy Ambramson
TMC

Junction Networks is the opposite of the companies mentioned in this MarketWatch article. Not only are we not imploding, but just this week we hired two new employees. You can look forward to speaking with Eric P. as he starts on the sales/technical support phones next week. Steve is a developer who was hired to help further increase the functionality of our product.

Since we are not VC funded, we are not living in fear that our VC will either pull their funding or not help us 'get to the next level.' Secondly, we are cash flow positive today and plan to stay that way. We did not start Junction Networks nearly five years ago to 'get rich quick' in any sort of dot.com or Internet 2.0 bubble. Our goal was to provide a real service with real value to real customers. We continue to strive to accomplish that every day. To our customers and staff who have helped to get us there, thank you.

--Mike

I agree with the article at "ZD Net about the current Money Crisis being good for Open Source. Red Hat's CEO, Jim Whitehurst, stated that the current economic climate would cause more companies to consider open-source software as an option.

Whitehurst said that this was because open-source software provided a better economic model for creating software. We, at Junction Networks agree: OpenSource, in any economic climate, is a better economic model.

I've also seen it in my personal life. My daughter's friend was throwing out a PC because she couldn't upgrade it. My daughter, who did not have a personal computer, decided to pick it up to see if she could do something with it. She (with some help from me) put Debian on it and, aside from some WiFi issues,it is working great. I would think we'd see something similar in the work-place as well. Instead of upgrading the hardware and software, I would expect to see firms start to shun the bloatware and switch existing hardware to OpenSource alternatives to keep from having to make further capital expenditures. That's why we love OpenSource.

There is a great post over at Bradley Holt's blog about using Google Talk's XMPP service in conjunction with SIP from Junction Network's OnSIP Hosted PBX.

Here is my take on XMPP and SIP. XMPP is an XML-based open standard with, among other things, built-in protocols for subscribe and publish, e.g. subscribe to someone's presence information and publish your own presence. SIP, another open standard, is good for a bunch of rapid-fire, short messages back and forth. It's kind of like DNS on steroids. SIP is currently (and for the foreseeable future) the de facto standard for VOIP.

Internally we have been running an XMPP server for presence and instant messaging for a few months now at our domain: junctionnetworks.com. Additionally, using open XMPP servers like Google Talk, I can see the presence of my XMPP 'buddies' at other domains as well. If you are connected to an XMPP server, you can chat with me at mike-at-junctionnetworks.com. That address is my e-mail, SIP and XMPP address. That's what I call real unified communications. We are making these XMPP services available to all of our customers at either their current onsip.com domain or even (eventually) at their own domain. (This is something we offer today for SIP calling.)

XMPP and SIP are two great tastes that taste great together. We have written an SIP to XMPP gateway which allows us to gateway SIP information to an XMPP server. Once you have that, you can then receive instant messages and screen pops on inbound calls. Our current project is to wrap all of this up in an easy to use web-based interface for the end user. One interface will handle your presence, chats and phone calls. We'll be writing more about that in the coming weeks. The point is that we are excited to see our customers looking to use the same technology and standards that we are building toward. As always, we will keep with our corporate themes of no walled gardens, open source and open standards.

There is a large Global Crossing (GLBX) outage in New York. It is affecting all customers coming to Junction Networks via the GLBX network. We are attempting to re-route traffic, but the issue is inside their network. They are still advertising routes to Junction Networks (and others) but not routing the traffic. We have escalated the issue with GLBX.

We are currently experiencing one-way audio on all inbound phone calls. Outbound calls are unaffected. We are currently tracking the issue.

ZD Net has the following article predicting the demise of SaaS in two years.

For those of you who do not know, SaaS is Software as a Service, a la Salesforce.com. We at Junction Networks use Salesforce.com and I love it. Anywhere I go, home office, NYC office, iPhone, I can get salesforce.com status. The same for gmail (who handles our corporate e-mail). It's the 'access it anywhere' model that makes SaaS such a success.

As computers become more and more mobile, think iPhone, the business user is going to want even more software as a service. I can't image having to load software on every device I may want to use across three or more different operating systems just to get at the information I need.

VOIP is increasingly becoming SaaS. Our own OnSIP Hosted PBX is one such example. Regardless of location, you can access your OnSIP account via a browser and the SIP phone calls are accessible via any SIP device anywhere.

The reasons SaaS will remain are two-fold. One, the internet is more capable to handle high bandwidth applications such as large data transfers and multi-media traffic like YouTube and VOIP. Secondly, devices are becoming ever more powerful, smaller and more portable. More bandwith plus mobile computing is a the perfect recipe for SaaS.

As a software-on-cd provider, are you really going to port your interface to the iPhone or the Zone or the Dare or whichever new mobile device comes to conquer the world? The answer is a SaaS model where the software lives in the cloud and is accessible via any WWW browser anywhere. The indicators for SaaS show no signs of stopping, so, in my opinion, the rumors of SaaS' demise are greatly exaggerated.

The OnSIP Hosted PBX has always had great voicemail features including voicemail to e-mail and voicemail notification via SMS.

With last evening's software release, we now offer the ability to transfer out of voicemail during the greeting. This is a feature that is specific to each voicemail box and must be set on each voicemail box individually. The functionality allows a caller to press "0" at any time during the greeting to transfer out of the voicemail box. In so doing, they will be transferred to wherever the user has set the "On Press of 0, send caller to:" setting on that voicemail box.

This feature allows the caller to try multiple extenstions within a company without having to hang up and re-dial. Typically, you would want to set the 'transfer to' feature back to the main auto attendant or back to the receptionist.

In order to take advantage of this feature, you must go into any existing voicemail box and update and save the setting before it will take effect.

Please Note: During the greeting, you can still hit the "*" key to log into voicemail and "#" to skip the greeting and go straight to leaving the message.

Note the "On Press of 0, send caller to:" setting on the below image: