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While a true story, the following contains what may very well be a bad idea...

I have an external 250 GB hard drive that, as of a few days ago, would not spin up. Upon power up, it would make an short buzzing sound following by, click, click, click, click, click and so on. I tried disconnecting and reconnected it a few dozen times with different cables, different ports, different power supply. No luck.

Finally, I decided to try freezing the drive by letting it sit on top of a large gel ice pack for 15 minutes. While still sitting on the ice pack, I powered it and it spun up on the first try! I then immediately began copying everything off the drive. The drive ran for about 3 hours while the ice pack slowly melted. Then it was back to click, click, click...

I got the idea from here:

http://www.macosxhints.com/article.php?story=2006110111270170

Lately, there's been a lot of buzz in the VoIP community about high-definition telephony. The first time we heard about it, we all raised our eyebrows since high-definition is a term that gets overused in technology. But we're open-minded geeks, so we tried it...and it's really, really cool. The audio has so little distortion compared to a traditional telephony call that it's slightly unnerving at first because the vocal clarity approaches stereo quality.

So what is it? High-definition telephony comes down to the G.722 codec. In pragmatic terms, the codec is responsible for the compression and decompression of the media stream (i.e., your voice) that comprises a VoIP phone call. One of the first things a SIP phone does when it talks to a SIP proxy is to negotiate which codecs both support so that when the media stream comes, it is compressed and decompressed at the same rate and overhead. G.722 is one of the most efficient codecs out there, so it does a very good job. Voila, high-definition.

(For more information on codecs, Voip-Info has a thorough list.)

The OnSIP Hosted PBX already has high-definition functionality for extension-to-extension dialing. Both parties will need a phone that supports G.722, such as the Polycom IP 560 or the SNOM 320. (Please note, the Grandstream 2000 has support for G.722, but it's problematic.)

But we won't stop there, of course - we're currently testing moving some of our other applications to high-definition. We'll announce their launch in this blog, so stay tuned!

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:

First, let’s address SIP: Session Initiation Protocol is the de facto standard protocol for establishing, conducting and ending a VoIP call. If you are really interested in technical details, check the Wikipedia entry on SIP.

SIP is to VoIP as SMTP is to E-mail. Just like the standard protocol, SMTP, which allows two email servers to exchange email data, SIP allows two endpoints (IP Phones) to connect to one another using a standard protocol. Without such a standard, two phones would not share a common set of instructions and guidelines in order to properly exchange packetized voice.

The great thing about SIP is that it promises to help do away with phone numbers all together. How? Imagine your email address could be “dialed” from a phone and it would kick off a call. Further, imagine everyone at your company is reachable by phone using email addresses. My name is Robert. It would be great if people could reach me by dialing Robert@xxxxxx rather than some random collection of digits, which have no meaning.

Think of SIP as the protocol for yet another service capability of a domain:

Email: SMTP
Web: HTTP
Voice: SIP
Video: SIP

OnSIP is a SIP service provider. With an OnSIP account, you can open your domain to SIP traffic. This will allow you and your team members to be reached by phone using email address too!

You can make and receive SIP calls to/from any SIP address. Calls to users on our network, since they are all SIP calls, are all free.

Check out our SIP Domain Features for more info.

It has been a very busy week with the Junction Networks developers. There are two new enhancements to the OnSIP Hosted PBX platform: External Phone Numbers in Groups and Call Details in Reports.

External Phone Numbers in Groups
Many of you have been asking for the ability to have external phone numbers (cell phones, home phones, etc.) as part of a Group. Now, with all Groups (found under the 'Groups' tab) you can have up to 10 addresses (users) in a group and up to three of them can be external phone numbers. This feature is now active and live in the OnSIP interface. With our $39.95 per month Hosted PBX plan, you get up to three Groups.

If you choose 'Simultaneous Ring' strategy, all users (and now external phone numbers) in the group are dialed at the same time. The first user to answer the call is handed the call. With the 'Hunt Group' strategy, calls are sent to each user individually up to the 'Ring Each for:' time limit.

With both group types, individual failover settings are over-ridden by the Group failover. That means that you can have a Group ring a set of users for up to two minutes and the call will not failover to any individual user's voicemail box because the group failover supersedes the user failover. However, this is not the case when there is an external phone number involved. If the external phone number 'answers' the call with voicemail before the Group failover timeout, then the external phone number will be given the call. Be sure to take this into account prior to assigning external phone numbers to groups.

Call Details
Until the most recent release, if you wanted to see individual call details you had to download the full Call Detail Record as a CSV (spreadsheet) file. Now, under the 'Reports' section if you click on any individual user, by default you still see their 'Summary' view, but you can now click on 'Show Detail' to see the last 25 calls made by that user. All of the inbound calls are still listed under the user listed as the 'Bill Calls To' user listed under the phone number itself under the 'Resources' tab.

We have a new voicemail feature slated for next week...

Some of you may have seen the press release from Microsoft announcing the release of the Response Point PBX and the role of Junction Networks as a preferred service provider.

Our favorite write-up of the partnership was over at Asterisk VOIP News, though that may be because they included a glowing note from the editor about the reliability of our service.

We tested the Response Point in our labs, of course, and found the setup to be almost ridiculously easy. The test network was set up and ready to go within 30 minutes of unpacking the box the system came in. Most of the Response Point customers that we've spoken to report a similar experience. The Junction Networks configuration can be downloaded from within the Response Point interface, which further simplifies the setup. Response Point is ideal for a 1 - 50 employee office with a set location and is geared to the non-technical administrator. The Response Point is a PBX, however, and the phones are "dumb", so they have to stay on the same LAN as the Response Point. If you like to move your phone around to locations all over the world, you'll probably be happier with our OnSIP Hosted PBX service.

We've put up a Knowledge Base article as well.

Many people ask me what a virtual phone number is. Well, everyone knows what a phone number is. So the real question is “What is Virtual about a Virtual Phone Number?”

First, here is a dramatically oversimplified history lesson:

Historically, phone numbers were tied to physical locations. The phone company would provision a phone number to work over a single physical line, which would be “dropped” at the actual location the number would be tied to. Calls to that number could only be delivered to that physical location and businesses would have to receive the calls using expensive PBX systems which maintained routing smarts, voicemail applications, IVRs, etc.

With a virtual phone number, the physical limitation is removed, allowing a company to use a phone number in a more flexible manner with no reliance on physical presence of phone lines or phone systems. Calls to a virtual phone number are handled by a remote agent or proxy, which forwards on calls based on user defined rules. This allows a business to:

• Seamlessly connect multiple locations
• Eliminate on-premises telco equipment, telco space, phone lines, etc.
• Maintain phone service during incidents effecting physical offices.

Here is an example of how a business uses a virtual phone number from OnSIP:

Company X maintains a New York headquarters and a Los Angeles sales office. The company has local phone numbers and one toll free number, all virtual phone numbers. When a customer calls the toll free number, rather than having it answered by a phone system in either the Los Angeles or New York office, an IVR answers the call on the OnSIP Virtual PBX service. When prompted by the IVR, the caller selects option 2, for sales. Because there are sales associates in both offices, phones ring simultaneously in both offices until answered in Los Angeles. The call is from a key customer who needs to speak to the CEO who is working from his beach house in Cape Cod. The sales associate transfers the call to the CEO who is connected to OnSIP using his home office cable Internet connection. When the call is completed, the CEO uses 4-digit dialing to a make a free call to the sales associate in Los Angeles to congratulate her on a job well done for helping close a major sale.

The entire team is connected via OnSIP, which acts on behalf of the users, no matter where they are now or where they move. Users have the flexibility to make and receive calls and use the service as if they were in the office at all times.

OnSIP has phone numbers available throughout the country and are available for immediate activation.

Did you know that Junction Networks has a public web service API available to any of our customers that exposes all of the pieces necessary to manage your own hosted PBX and PSTN gateway services? In fact our admin.onsip.com web administration portal has been built entirely on top of the very same API that is open and available to the public - this means any feature you see in our administration portal is potentially available for you to implement in your very own VoIP product.

What does this mean for your business?

This means that you could build your own web portal to extend your product line to include a VoIP service without having to do the work of building the VoIP side of the service! You would be able to leverage all of the features from our OnSIP Hosted PBX and PSTN gateway products into your own product behind your own custom web portal. For example, let's say that you are a web host looking to augment your product offering in order to differentiate yourself from your competitors. By implementing the various pieces of the Junction Networks web service API you would allow your own customers to create and manage their own hosted PBX. You could even do some more advanced things like host your customers SIP domain in the primary domain that he has registered with your service already. Simply add an SRV record to identify Junction Networks as the domain's SIP service provider and the customer will be able to get their web, email, and voice service all from you - while you don't need to worry about any of the voice end of things. Additionally, you can implement your own pricing structure on top to fit your new VoIP offering to be in line with your current pricing. Since OnSIP never charges for users or extensions you can offer your implementation either with or without such charges - the choice is yours.

The benefits of implementing your new VoIP product offering this way is that you can continue to concentrate on your core business without a need to dedicate permanent valuable resources to building and maintaining your VoIP product. That's why we're here! It is our job to make sure that your VoIP services continue to stay up and working reliably.

Getting Started with the web service API

To get started with the Junction Networks API you can start by trying it out yourself on our API demo page. In order to use the demo you need to be a registered Junction Networks customer with an authentication name and password, you can signup for a free 30 day trial here to obtain these credentials. You can also get started with our VoIP web service API by reading the API documentation. If you happen to be looking for a feature that is not currently documented then please send us a support request and we'll be sure to get the API call documented as soon as possible.

Have fun.

We've had a really busy week in the Junction Networks lab. We've started a testing program for some of the routers out on the market so that we can come up with some solid recommendations for routers that are known to work well with our network. (Alas, not all routers are created equally.)

The main rule is that the router needs to not interfere with SIP. Specifically, we need routers that don't rewrite SIP packets, because we do a fair amount of work in learning about the network that a packet comes from in order to route and transfer calls properly. When a router rewrites the packet its sending us, we start to see problems, because it's essentially lying to us about the network behind it.

One trend that we're seeing with router/firewalls (which for the SMB market are usually the same device) is the introduction of the Application Level Gateway. An Application Level Gateway does what it sounds like it does - it rewrites packets based on the application and hides the NAT, with the idea that this helps certain programs work with fewer problems. Unfortunately, in all the of the ALGs that we've seen so far, SIP packet rewriting is turned on by default. This means that we can't properly detect that NAT that the phones behind the router are on, which causes problems with call transfers.

One dead giveaway of this behavior for our OnSIP Hosted PBX customers is in the admin interface. If you look at the phone registration for a user, you'll see "NAT not detected" in red letters. This is almost always caused by an ALG.

The solution is to configure your firewall to remove the ALG for SIP. We're going through a series of routers in our testing and have found only one router that didn't have this functionality so far, but we resolved it via a firmware upgrade. Keep your eyes peeled to our Compatibility Guide for more information. We'll be updating it in the coming weeks with more details. We even have a new section in our knowledge base with information on how to turn off the ALG for routers that have made it through our labs.

Most days I work out of my home office near Philadelphia on a 5 year old Windows 2000 machine. Most of what we do at Junction Networks is virtual. We use Salesforce.com for lead tracking and trouble tickets and Google for document sharing and e-mail. I tried an experiment this week to see how virtual I could be and see if anyone would notice.

I bought a new MacBook Air. My first Mac. I have to say I LOVE IT. Buy Apple stock now. I copied over usersnames/passwords and bookmarks from my Windows box to my new Firefox install on the Mac and left town. I drove 600+ away on a 'working vacation' in Indianapolis (where I grew up and my family still resides) so my kids could hang out with their cousins for a week. I only brought with me my Mac and my Polycom VOIP phone. I wondered if anyone would notice. I didn't tell anyone what I was up to.

First problem: SSH. We use SSH to log into our servers and I didn't have my key on my Mac. I created a new key and e-mailed it to John and asked him to upload it to my .ssh directory so I could log in. I thought my cover was going to be blown right there. But I explained it as 'away from the PC' and trying to log in with the MAC. John got me hooked up and I was logged into our servers. Phew. Over the first hurdle.

Plugged the Polycom phone into the DSL line and it came right up. My username and extensions all 100% intact. I called ext. 7008 to talk to Tim in Chicago and it worked perfectly. The MacBook Air effortlessly connected to the WiFi connection and e-mail was up and running.

I have to say the week went well. I was 100% on the Mac except for one instance where I needed to manipulate an Excel file with some serious string concat functions that aren't in Google docs yet. Other than that, eight hours a day on the MacBook keyboard (including this not-short post) and there were no issues. It is an amazingly comfortable keyboard and easy to touch type on. It didn't crash once and Firefox was lightning fast even with all the tabs open.

I think I made it. No customers or other employees know I'm not in Philadelphia. I was 100% effective here being able to access servers, customer records, e-mail and documents without issue and with the phone, I was accessible via my extension and was logged into both the sales and support phone queue the entire week.

Tomorrow, on the weekly conf. call on Friday, I'll spring it on everyone that I'm 12 hours away from our NYC office, not 1 hour as they assumed. I'm actually pretty shocked that it turned out this well, but I'm sure for our engineering team it will be a case of, 'yeah, we planned it that way.' Which, of course, they did.

I'm not sure how many people can do this in their day job, but I'm happy that I can. I can 'secretly replace' the 'near' me with a 'far away' me and no one notices. Pretty cool.

Now I need to go on vacation where I leave all this stuff at home...