Business Internet Phone Service Blog
Junction Networks receives a mention in an article in Processor.com, a newsletter for the Data Center marketplace.
- mike's blog
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Why we don't run our systems "In the Cloud"
I occasionally get questioned about our choice to run our own network and servers when we could outsource it all to a "cloud" provider like Amazon and thus save a lot of money while simultaneously improving the reliability of our service. I hate this question because it is loaded, and I know they are not likely to understand why we can't run on a cloud service no matter how patiently I try to explain the technical issues.
Anyway, Amazon's EC2 cloud systems are now apparently dropping packets and having network latency issues. People running near real-time applications like gaming and VoIP are not having a happy time and they are apparently scrambling all around trying to figure out why their services are going to crap and, all the while, Amazon says there is no problem. I imagine that these folks did not get an SLA from Amazon with respect to network performance.
Whocouldaknown?
Link: Are spot instances killing the performance of Amazon EC2? - Seldo.com
- john's blog
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My email address is tim@junctionnetworks.com. That is also my SIP address. If you are using the my.onsip web application you can enter my SIP address into the dialing field and call me directly. This is SIP to SIP calling. The call quality is as good as, or better, than traditional PSTN calling and, best of all, its free. You can dial SIP addresses anywhere in the world and the calls are free, regardless of whether the person you are calling is using OnSIP.
So what are SIP addresses and why do they matter? Everything with OnSIP is a SIP address. My phone is registered as a SIP address (tim@junctionnetworks.com), my VM box has its own SIP address (vm.tim@junctionnetworks.com). The ACD Sales queue I log into has a SIP address (acd.sales@junctionnetworks.com). You can dial any one of those addresses from a SIP phone directly and avoid paying PSTN minutes charges.
You all have SIP addresses as well. Most of our customers have a user SIP address in the following format: joepublic@example.onsip.com (where the "example" in the domain name is the username under which the account was created). This is fine for most of our customers, as they rarely use or publish their SIP address. I think this is a bit of a shame. SIP addresses are an easy and very flexible way of communicating with other SIP addresses to make free calls. OnSIP wants to encourage you to use SIP addresses more and we offer SIP Domain hosting as a way to do this. You can have a SIP address that is the same as your email address and then have one less piece of contact info for people to remember.
I own the domain name sipmaven.com and I'm going to use that as an example to show you how I go about setting up domain name hosting with OnSIP.
Here is the user information for my new OnSIP account. My SIP address is tim@sipmaven.onsip.com and my goal is to change it to tim@sipmaven.com.

OnSIP is unique in VoIP service providers in offering SIP domain hosting but some of the leg work has to be done by modifying the DNS records of the domain name. The sipmaven.com domain name is hosted by GoDaddy so I'll need to log into their admin interface and make some changes to my DNS records. Specifically I'll be changing the SRV (service) records.
When I log into the GoDaddy domain manager I can see a number DNS records for sipmaven.com. I'm interested in adding a new SRV record.

I need to insert the following information into the various fields:
- Service: sip
- Protocol: udp
- Name: sipmaven.com
- Priority: 0
- Weight: 0
- Port: 5060
- Target: sip.onsip.com
- TTL: 1 hour

When I have saved my SRV the DNS page will look like this:

We'll have to wait up to 24 hours for the SRV changes to take effect and to propagate through the internet before we can go ahead and set up the SIP Domain hosting with OnSIP.
Next log into your admin.onsip.com site and go to the account tab. You'll see you have the option to Migrate SIP Domain. Select that and then use the pull down menu to use your own private domain and enter it in the open field. Save.

Please note that changing your SIP domain will effect all the SIP addresses in your domain.

Specifically, any phones that were registered at sipmaven.onsip.com will cease to be registered and you will need to change the proxy/domain setting to sipmaven.com before the phones will reregister. If you are using any SIP addresses for services like the Inbound Bridge they will also need to be modified.
Now you have your domain name as your SIP domain.
As we add more features to the my.onsip application you'll be able to take advantage of instant messaging, presence, click-to-call etc. using your domain name. This will make communications easier and cheaper for you and your customers.
More information on SIP domain hosting can be found here.
We also have information on modifying SRV records if you're using Register.com or No-IP.com as domain hosts.
- tim's blog
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Our Thoughts on the Packet8 Service Outage Comments: 1
Service outages happen. Sometimes it’s the result of something you did, but more often than not it’s caused by factors completely out of your control. If you’re in the business of providing a hosted service over the Internet, then chances are that this unfortunate situation will happen to you at one point or another. We would all love to say that it has been XX months or X years since we’ve had any problems, but the fact of the matter is that when something like this happens (and it most likely will), what’s most important is how you deal with it.
Now we don’t know exactly what occurred but from what we’ve heard and read (there’s quite a bit of chatter on Twitter from actual packet8 customers), it seems like Packet8’s customer service/response during the outage was a bit iffy.
We had a little outage ourselves last year and we dealt with it by putting up multiple posts on our blog, posting regular updates to keep our customers informed on Twitter, and responding quickly to any questions. It was vital for us to keep up a constant stream of communication with our customers to let them know that we were doing everything in our power to get things back up and running. In this day and age, there’s little to no excuse for leaving your customers high and dry, especially when they need you the most.
After our outage, we put up an explanation of exactly what happened and everything we were going to do to prevent it from ever happening again. Industry analyst Gary Kim specifically mentioned us on his blog in a post entitled, ‘The Way to Deal with an Outage'.
I’m not going to sit here and tell all the Packet8 customers reading this that we’re going to be outage-free for the next 10 years. We can’t promise that. No provider can.
What I will say is that as a company, we’ve always prided ourselves on our customer service and that will remain the same, whether everything’s running at 100% or not.
- leo's blog
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OnSIP is Google Voice for Business Comments: 1
e-Week is reporting that Google has big plans for VoIP in 2010. This is great news. Really great news. SIP-enabling Google Voice a la Gizmo5 would be a huge shot in the arm for SIP, solidifying its appeal to the masses. However, it still does not address the needs of business customers. Small and medium businesses are flocking to cloud computing in general and hosted VoIP specifically, but Google Voice is ignoring this segment of the market completely. Here are the top three reasons a small business should use the OnSIP Hosted PBX by Junction Networks instead of Google Voice:
Extension Dialing
SIP addresses are great. Mine is sip:mike@junctionnetworks.com. Put that address into your phone and my phone rings. Very cool. The problem is that business desk phones are still number pad based. If I want to call another Junction Networks user, it is far easier to dial their extension - which our system translates into a SIP address behind the scenes - than it is to type their SIP address into the phone via the number pad. For a business to use a VoIP system as an internal PBX, you must have extension dialing. Google Voice has no support for extensions. Currently, to reach a Google Voice customer you must dial their phone number. With the integration with Gizmo, that will expand to SIP addresses, but still no extensions.
PBX Functionality
If you have only one or two people in your company, then Google Voice is a good choice; mainly due to its cheapness. However, once you have five or more, you'll need services like attendant menus ("Hello, welcome to Acme Corp, press 1 for sales and 2 for customer service."), voicemail boxes, and dial-by-name directories. None of these features are currently supported with Google Voice. By contrast, the SoHo package of the OnSIP Hosted PBX includes three Attendant Menus, three ring groups, five voicemail boxes and one dial-by-name directory. That's everything a small business needs to get started.
Integration with Google Voice
Most VoIP providers make their service a walled garden. You may or may not be able to call outside SIP addresses, but very, very few allow you to add external SIP address to the PBX. OnSIP is different. We allow you to put external SIP addresses into your PBX and fully integrate them, even to the point of giving them extensions and using them in applications like the attendant menu. In this way, you can integrate Google Voice users into your corporate PBX. For example, if you have freelance developers with Google Voice accounts and SIP addresses, you can add them as extensions on your OnSIP Hosted PBX and get the best of both worlds.
To be fair, Google Voice is not targeting the small and medium business. Their purchase of Gizmo5 clearly shows that they are going after the Skype market. But the fact that they will be giving out SIP address further validates our reliance on a full integration of SIP. We are always happy to see big players adopt SIP.
- mike's blog
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OnSIP Now Supports High Definition Voice
Actually, we always did.
Is this really a big deal? At OnSIP, it’s a split decision.
If you ask our CTO, he’ll say, ‘What’s the big deal?’ We’re just allowing the phones to do what they are made to do. If you ask our marketing folks, they’ll tell you “Fantastic, HD voice is really cutting edge stuff for a small business at a groundbreaking low price!”
Here’s the deal. Phones from Polycom, Cisco, Linksys, SNOM, Aastra and countless others currently come with support for HD voice, meaning they support calls in HD using some flavor of the G.722 codec, the “wideband codec”. For two people to have a conversation in HD, other than both having HD-capable phones, the only requirement is for the provider completing the call to allow two endpoints to speak to one another in the native G.722 codec. A provider would literally have to get in the way of the phones, preventing them from doing what they are inherently capable of doing in order to prevent an HD call. That’s why our CTO and others here say “big deal” to countless press releases from providers boasting about an all HD VoIP Network.
Because of our strict adherence to the SIP standard, OnSIP has supported HD calls since the first day HD-capable phones were available. But, as our marketing team will tell you, it’s been a long time coming that we are boasting about it.
Why? Our marketing folks felt we should boast about it when it truly was groundbreaking. And now, with HD conferencing, we really are confident that our support for HD is truly groundbreaking. We are confident you will find no other provider offering the following:
HD calls between:
- Extensions – Inside your OnSIP account
- Customers – Outside of your OnSIP account to any other OnSIP user
- Internet Callers – Outside of OnSIP to any other SIP address
- Conference Bridge Callers – Any caller to an OnSIP bridge, using an HD-capable phone and SIP address
Using:
- Any HD-capable SIP endpoint (phone or software phone)
Yes, there are providers out there supporting the individual pieces, such as HD conferencing or extension calls using only a select few Aastra phone, but OnSIP is the first to support all of these features on any HD-phone as part of a complete business phone system at under $20 per user.
Should you care? Well, HD calls really do sound much clearer than anything on the PSTN. It’s a huge upgrade and we recommend our customers buy HD-capable phones to take advantage of it.
BTW, we also support ultra-wideband voice, HD video…..on any device that supports it. :-)
- rob's blog
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OnSIP's 1st Holiday Twitter Giveaway is Underway

It's the time of year for giving! This year, the OnSIP team is getting into the holiday spirit by hosting our first holiday giveaway.
Enter on Twitter and you could be one of the lucky ones starting 2010 with a brand new Netbook. Tweet us an interesting story about your business phone or phone system and you'll double your chances of winning.
For more information and the full contest rules, visit our contest page. Good luck!
- leo's blog
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OnSIP Tips from the SIP Maven: Announcements
What is an announcement? The dictionary definition is: "A broadcast message, especially a program note or commercial" which sums up nicely what the OnSIP accouncement application does.
Each OnSIP package comes with one or more announcements and additional announcements can be purchased for $4.95 each per month.
Most OnSIP clients use the announcement to broadcast some company information by linking to it off the main attendant menu: "Please press 4 for the company fax number", or "press 5 for our mailing address." Then, when the caller presses 4 they hear the announcement recording: "Our fax number is 212-555-2345." After hearing the announcement the caller is sent back to the main menu.
But you can do more with an announcement. How about letting your callers hear about your December promotions before they get transferred to the sales queue or group? Do you want to let callers know about your Holiday hours prior to going to your main attendant menu? It's all easy to do.
To add an announcement for an upcoming December sales promo go to the Apps tab and click on the Create New Application link.

You will need to add the specific announcement recording .wav via the Resources tab.

Then go and modify the attendant menu you want the announcement to play off of. In the attendant menu shown:

I have changed the destination of "On press of 1:" to: The December promo announcement. You'll see in the next picture that I have selected the Transfer To: The Sales group.

So, when a caller dials my number they go to the main attendant menu, choose "1 for sales" and hear my December promotion before getting transferred to the The Sales Group.
You can record multiple announcement messages using the Recording resource and substitute them for each other depending on specific needs. You'll find that announcements are an easy and cost effective way to broadcast messages, especially a program note or commercial.
- tim's blog
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OnSIP is Polycom VoIP Field Verified Partner
The team here at Junction Networks is proud to say OnSIP has recently become one of Polycom’s VoIP Field Verified (VFV) Partners. Polycom’s VFV Program “enables call control providers who have successfully integrated with Polycom’s VOIP family of phones to complete certification and resell Polycom phones.”
We’re very excited about this partnership. In our experience, Polycom phones have always been top notch and we are honored to be working with Polycom. Over the years, we have had consistent feedback from customers telling us that Polycom is a top choice vendor of IP phones.
Businesses can expect a superior experience when using Polycom phones with OnSIP. And to further OnSIP’s support of Polycom hardware, we recently released our new boot server to make provisioning phones and updating firmware easier for our customers.
Shortly, our customers will be able to buy Polycom phones preconfigured from trusted IP telephony vendors essentially eliminating the time and work required to provision phones.
Polycom was one of the first manufacturers committed to bringing wideband audio support to the IP telephony industry with their ‘HD Voice’ offerings. In accordance with our dedication to seeing wideband audio widely implemented, today, OnSIP customers can call one another for free in full HD. Furthermore, the OnSIP team will soon be rolling out greater ‘HD’ support. Be sure to look for our HD conference bridges in the near future.
- leo's blog
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The True Cost of Free Conference Calling Comments: 3
There is no easy way to say this. Starting Friday November 13, 2009, calls to these seemingly free conference services and other reverse billing services will be charged at $0.50 per minute.
Starting Friday November 13, 2009, the affected rate centers are:
(712) 432-xxxx and (712) 338-xxxx
The Problem:
Calls to these rate centers are 20 times more expensive than a ‘normal’ call. Junction Networks cannot afford to subsidize these services and at the same time maintain our competitive pricing. We have only two options available to us – block calls to those numbers or charge the true market rate. We have chosen the latter.
The exposure of the reverse billing services has been in the news quite a bit lately. Some carriers have chosen to simply block these numbers. Speakeasy has an extensive list. Even Google has apparently noticed the same issue.
How do I turn on/off the ability to make these calls?
Junction Networks customers have the ability to turn on/off access to any call costing more than $0.029/min. Currently, unless you have filled out our Extended Dialing Form, access to more costly calls is turned off by default.
As we do more analysis of our bill, we expect to fine-tune the affected rate centers. Please see our full pricing schedule.
We know this will affect a number of our customers, as well as Junction Networks, as even we have been using these ‘free’ services. We expect that, as other carriers are handed large, unexpected bills, they will also be forced to pass on the true costs to their customers, or simply block access to these services. Eventually, the entire carrier compensation program that has been in place for decades will be challenged and likely overhauled, thereby ending the loophole that has allowed ‘free’ conference calling to exist. If and when rates to these calling areas return to normal, we will return to our standard $0.029/min for the affected rate centers.
As always, we appreciate your business and we do sincerely apologize for any hardship this may create. However, for Junction Networks, the economics are unsustainable.
- mike's blog
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