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OnSIP

Mar
08
2010

OnSIP Tips from the SIP Maven: Customer Q&A

Posted by tim at 03:59 PM EST

Question: I’m going to be traveling in the next week. Can I plug my Polycom phone into my computer and use a Wi-Fi network to make calls?

Answer: The short answer is no. Most VoIP phones have a simple switch that enables a computer to connect via Ethernet to a phone. You can’t reverse this process and connect a phone to a computer.

There are a few alternatives. You can use a softphone on your laptop and make and receive calls using a headset. Our favorite softphones are made by Counterpath. We have detailed instructions on configuring X-Lite and eyeBeam.

If you have an iPhone or Android OS smartphone you can use a SIP app to install your OnSIP user. Fring and iSIP work well.

If you are going to be staying somewhere where you can add a device to a Wi-Fi network you can also use an Apple Airport Express and plug your SIP desktop phone directly into that. The Airport Express connects to a WiFi broadband router that you have access to (so its not going to work in a Starbucks or a hotel, but its great if you are going to the parents or in-laws).

You can plug the Airport Extreme into any power outlet and then run your SIP phone from there. I use my Airport Express/Polycom combo to work in different rooms around the house (or even outside in the summer).

Mar
01
2010

Celebrating the 1-Year Anniversary of the OnSIP Agent Program Comments: 1

Posted by tim at 12:51 PM EST

The OnSIP hosted PBX agent program recently celebrated its 1st anniversary and has just added its 75th partner. This incredible number says a lot about the energy and determination of Andy Ogg, our agent program manager. It also says plenty about the success our agents are having at signing up customers and making OnSIP hosted PBX a part of their monthly recurring revenue stream.

So what is the profile of a thriving OnSIP Hosted PBX agent?

Agents are typically part of smaller companies, in the 2 to 20 range. They often specialize in providing IT services to the SMB marketplace in a particular geographic area. Selling a VoIP service is a natural progression from selling LAN/WAN products.

Some agents are deploying vertical strategies, targeting industries such as healthcare, real estate, education, religious organizations, and not-for-profits. Vertical selling is not limited by geography and agents can ship preconfigured hardware anywhere in the country.

Not all of our agents come from a traditional network or telecom services background. OnSIP is more than just a phone system; because we utilize the SIP protocol, we provide a unified communications platform. Because presence, instant messaging, click-to-call, and video conferencing are all included with the OnSIP hosted PBX, we are seeing media consultants and web production companies signing up to be agents. When your SIP address is the same as your IM and email address, you can significantly simplify communications.

A final group of OnSIP agents are equipment and hardware vendors. Hardware margins are minimal and it can be difficult to expand your business with such slim margins. However, by combining hardware phone sales with a customer install and then including monthly recurring revenues, you can grow your business profitably.

Why is the OnSIP VoIP reseller program successful?

Agents sell OnSIP for three reasons:

  • OnSIP is an easy and reliable business VoIP solution to provide customers
  • The monthly recurring revenue grows as agents add customer
  • To provide a full range of data and communications services to their customers

We’ve intentionally made the OnSIP agent program as easy to sign up for as OnSIP is to set up and install. We do have an agent agreement, but we don’t have any customer account minimums, exclusivity, or scary NDA’s. We take care of the billing and provide the tech support.

OnSIP agents keep their customer by providing a hosted VoIP service. If they don't offer their customers VoIP, another vendor will.

Please contact Andy Ogg directly to find out how to become an OnSIP agent.

Jan
25
2010

OnSIP Tips from the SIP Maven: SIP Domain Hosting

Posted by tim at 12:13 PM EST

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.

Jan
06
2010

OnSIP Now Supports High Definition Voice

Posted by rob at 02:01 PM EST

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. :-)

Dec
17
2009

OnSIP's 1st Holiday Twitter Giveaway is Underway

Posted by leo at 01:50 PM EST

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!

Dec
08
2009

OnSIP Tips from the SIP Maven: Announcements

Posted by tim at 12:24 PM EST

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.

Dec
03
2009

OnSIP is Polycom VoIP Field Verified Partner

Posted by leo at 02:04 PM EST

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.

Oct
16
2009

OnSIP Tips from the SIP Maven: External Numbers

Posted by tim at 04:07 PM EST

Managing External Number Resources

By adding cell phone and international office numbers to OnSIP you increase the usefulness of your PBX. Calls can be routed to a cell phone rather than to voicemail so you never need miss a call. You can also set up alias extensions to dial out to external numbers. This makes getting in touch with colleagues who are out of the office as easy as dialing an extension. Make things easy by assigning cell phones extensions to be one order higher than regular extensions; my work extension is 7008 while my cell phone extension is 8008.

When you add a new external phone number resource we list that number in appropriate pull-down menus so as to make it available as a new extension destination. However, by default, we just list the the phone number. Have a look at this screen shot listing 13 external phone numbers.

Unless you are really good at remembering peoples numbers you are going to have a hard time matching cell phones to users. By clicking on each number and then selecting modify you can edit each number and give it a meaningful name.

Then save.

If you do this for all your numbers you can easily assign a cell phone or other external number to correct user or extension.

Sep
21
2009

VON Show - Day 1

Posted by mike at 12:52 PM EST

Junction Networks is at the VON Show in lovely Miami Beach today and tomorrow.

This morning I moderated a panel titled "Beyond Boxes: The Future of the PBX." My panelists were Dan York from Voxeo and Chris Thompson from Adtran. It was a great session. We actually ran out of time. One hour was not enough. The panelists were very interactive and made it a breeze to moderate.

First meeting was Standing Room Only.

The discussion centered around Hosted PBX (AKA Voice in the Cloud) vs. On-Premise PBX. Adtran manufacturers on-premise PBXs and was leaning that direction. Voxeo and Junction Networks with our OnSIP Hosted PBX were leaning toward Voice-in-the-cloud services. Both sides, however, admitted that there were times where one type of PBX rose above the other and even times when a hybrid solution was best.

The second session was called "What's in a Number The Future of Dialing and Identifiers in the Mobile Universe." Panelists included Rod Ullens de Schooten co-founder and CEO of Voxbone and Joel Fisher, VP of Marketing for Telcordia.

Second session grew to nearly full by the end. It started at about 3/4 full, but grew over the hour.

Personally, I would like nothing better than for phone numbers to disappear all together. If a phone number I want to dial is not on speed dial, I'm not calling it. There are just too many numbers. SIP addresses. That's the way to go. My wife's company recently started using Junction Networks. I no longer had to have her main number on speed dial and then dial her extension. Now, she had a SIP address. I type in her address and I'm calling directly phone to phone. I love it.

The consensus of the panel, however, was that phone numbers will be around for quite some time. 20+ years seemed to be the general thought. I've been hoping for the last five years that faxing would die a quick death, but no such luck. The same appears to be true for phone numbers. Junction Networks, however, will keep doing as much as we can to promote SIP addresses and SIP interoperability.

So far the show is doing well. The show floor just opened, so I'm back to work.

Sep
04
2009

Skype for Onsip - Would you use it? Comments: 2

Posted by mike at 03:35 PM EST

Digium just announced Skype for Asterisk. If we created a Skype gateway for OnSIP would you use it? Comment below or send e-mail directly to Mike.

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