OnSIP blogs

Unlock Your Google Voice Phone Number and Set it Free

matt — Thu, 09 Feb 2012 17:40:27 +0000

By Mike Steiner and Matt Barrett

It has become common for people to feel attached to their phone numbers these days. Whether it’s been printed on your business cards or is saved in everyone’s address books, it can be a pain to change numbers. Perhaps that's why so many people decide to port when switching carriers. In order to successfully port, however, it is important to know a little bit about how your carrier handles this process. In this example, we feature Google Voice.

We've encountered a few instances where customers previously leveraged Google Voice to handle their business phone service for just one or a few users. For, say, two business partners, Google Voice is a nice solution - offering a free number for each user, free voicemail transcription and management, free SMS texting, and conferencing for up to 4 callers. When their businesses grew to 5+ users, however, they found they needed further PBX services such as a main phone number, auto attendant, extension dialing, conferencing for more callers, call routing rules, etc., plus the ability to use a desk and/or conference phone. Enter OnSIP. While we don't offer a few of the GV single-user perks (e.g. SMS texting, whisper app), we have a variety of applications you would expect from a business phone service, and users can choose from a variety of IP phones.

Until recently, a few of our customers had difficulty bringing their Google Voice numbers to OnSIP. For instance, if a number has not been unlocked with Google, all porting requests will be rejected, prolonging the process of switching carriers. This has occurred even in cases where our customers have provided all of the other paperwork. Fortunately, we have found out how to unlock this number; if you have tried porting a Google voice number to us before and were unsuccessful, you may want to submit a new port request after reviewing the steps in this blog.

The Google Voice website offers some basic guidelines to make the move easier. Of course, the most important thing to note is that Google Voice, unlike most carriers, charges a $3 unlocking fee. While this is a bit of a nuisance, we have found very few problems when attempting to port from GV to us after the number has been unlocked.

Once your number has been unlocked, you can follow the normal porting procedures as found in our Knowledgebase. In place of a phone bill, however, you can log into your Google Voice account and capture/print information that includes the billing name, billing address and phone number to be ported. Most online portals have a billing section with your address information for credit card billing. Along with the paperwork containing your name and address, also provide the unlocking-fee receipt in order to ensure this process is completed as quickly as possible. From that point on, you’ll be ready to port!

Once your number has successfully ported and is working as desired, it's always good practice to notify the losing carrier (in this case, Google Voice) and inform them that you've ported your number out. You should also request that they remove your number from their system. Otherwise, it is possible that current customers of the losing carrier will be unable to reach you at your new destination.

So what are you waiting for? Businesses, bring your numbers to OnSIP today!

Reports of inability to make calls and access admin.onsip

nicole — Tue, 07 Feb 2012 23:12:08 +0000

6:15 Network Alert
We are aware of many customer reports of an inability to make calls and access admin.onsip.com as of 6:10 PM ET. We are treating this situation with the highest priority and will update this network alert with more information as we have it.

6:31 ET Update
Services appear to be back for some customers. Please feel free to update us here with your status. We will add more RFO detail shortly.

6:45 ET Update
Services were up briefly at 6:30 ET, but we have confirmed network issues in our NY datacenter at this time. We greatly apologize for this interruption in service. We are working to resolve this situation and will continue to update.

7:20 ET Update
We have engineers on-site in our NYC datacenter and should be able to update with more specific information soon. Once again, we greatly apologize for this inconvenience and understand this is causing an interruption in your work day. Please continue to refer to this Network Alert.

7:38 ET Update
Services should be restored for our customers. We diagnosed a hardware failure in our NYC datacenter and have made the appropriate replacements. We are reviewing our service monitoring. Please update us if you are still having problems. We will continue to update this network alert.

8:10 ET Update
We have confirmed network stability. We will will follow up with an official Reason For Outage and our remediation. Thanks to everyone for your patience and communication.

February 8, 2012 Postmortem Report
On February 8, 2012 at approximately 6:16 PM ET, all OnSIP services except SIP to SIP calling became unavailable. Our Engineering staff was dispatched to our NYC datacenter and identified the problem to be a hardware failure. More specifically, a switch failed and caused critical services to lose connection to our network. In this situation, our redundancy measures failed, but the Engineering team replaced the hardware, bringing services back up at approximately 7:36 PM ET. After monitoring our systems and receiving customer feedback, we reported full network stability at 8:10 PM ET. In the long term, our Engineering staff has identified the issue and is making improvements to our network redundancy to prevent this cause of service interruption from reocurring.

We would like to reiterate to our customers that we're sorry for this service interruption and understand the upset to your work day. We know our phone service is a critical piece to your business. We are grateful to our customers and the constructive feedback we received during this time. In particular, we are going to make the following improvement as a result of your suggestions:

Some customers expressed difficulty in finding our network alerts. To provide you with up-to-date, accurate information, we like to put our real-time communication in one spot - our network alerts. To ensure you can find information you need sooner, we are planning for a website redesign to be launched in March, and we will post network alerts prominently to the homepage.

Thank You,
The OnSIP Team

The Travails of Caller ID

nicole — Mon, 06 Feb 2012 18:23:07 +0000

Co-written by Larry Browne, member of the OnSIP support team, and Nicole Hayward, member of the OnSIP marketing team.

We get many calls and cases in the OnSIP support bunker about caller ID. The most common one is “Why isn’t my caller ID showing when I call my wife?”

Yeah, about that... got a few minutes? I won’t completely ruin your day with the long, technical (boring) explanation. Here's a quick explanation about the the history of caller ID and OnSIP's implementation of this feature.

The History of Caller ID

Caller ID started as, and still is, a vanity feature. "The first market trial for Caller ID and other "Custom Local Area Signaling Services" was conducted by BellSouth as one of the "TouchStar" services on July 7, 1984 in Orlando, Florida. The Lines of Business (marketing) department in BellSouth Services named the service 'Caller ID'" (Wikipedia)

Caller ID is actually two bits of information: the caller's number and the subscriber name associated with that caller's number. This information is transferred as so: When a call is made, the originating phone switch sends the caller's number. Then, with the caller's number, the callee's service provider is responsible for looking up the caller's subscriber name.

When it was developed, the world was primarily a realm of Plain Old Telephone Service (POTS) lines, and the caller information was tied to the Central Office switch to which the wires were connected. Since there were also relatively few carriers at the time, it was easy to keep track of the caller ID information in one place.

Caller ID Today

Enter wireless and VoIP phones, hundreds of local/long distance carriers, Baby Bell breakups, consolidations, mergers, and the like, and the map has gotten much more fragmented. Unlike phone numbers and DNS, wherein there are internationally recognized databases that are authoritative sources, there is no central authority or regulation for caller ID. There are no FCC guidelines regarding its accuracy by carriers. While there are federal regulations regarding telemarketers and spoofing, carriers can maintain their caller ID databases as they see fit.

We offer caller ID for our OnSIP customers, and we don’t charge extra for it. We offer this feature by querying a reputable Caller Name (CNAM) database.

How OnSIP's Caller ID Works

On an inbound call, the call is sent to us, often from the Public Switched Telephone Network (PSTN), with a 10 digit number sent from the caller's provider. We take that number and look it up in the CNAM database to which we subscribe, and if there is a name, we send it to the OnSIP callee's registered phone(s) for display. Usually, this is between 9-12 characters. If there is no information on file, then we will send a best approximation of the geographic area for the phone number (e.g. New York, NY).

Outbound caller ID works the same way: we send the number to the PSTN, and the callee's provider does the lookup against their database, which may not be the same as ours. If the callee's provider uses the same CNAM database as we do, then chances are the caller ID name you have set in your account or user record will accurately appear there.

We send all subscriber updates that OnSIP users make to our CNAM provider every night, and the changes can show up quickly - in as little as 24 hours. However, the information may take a while to propagate to other CNAM databases. So, if an OnSIP customer updates their Caller ID settings, the change will be quickly evident if s/he calls someone whose telephone service provider uses the same CNAM database. Unfortunately, if the callee's provider uses a different CNAM database, the change may not be evident for weeks.

When a call is made to a cell phone, most times the phone does a lookup against the contacts in the phone’s address book, and if the number isn’t there, will display Unknown. This varies across providers, so check the service level of the individual provider for exact information.

Final Note

This is the long and short of caller ID. Because the other providers use different CNAM databases, and some rarely or never update them, the information received on the termination end is sometime wrong. And since it’s a feature, and not a regulatory requirement, they rarely have the time or inclination to fix it.

The good news is that we use a top, reputable CNAM database, so the caller ID information passed to you is dependable. I hope this has been a brief and helpful overview of caller ID!

Testing Paging Systems for OnSIP's New Call Park Feature (Coming Soon)

sukanya — Fri, 03 Feb 2012 18:58:11 +0000

This blog is by Eric Phipps, NOC Technician

One of the long requested features we’ve had at Junction Networks has been the introduction of a parking server, which would allow us to place a call on hold and allow another phone to pick up that call. We will shortly begin rolling out this feature, but one of the questions we dealt with while developing the feature is: what’s the easiest way to let people know that they have that call? We think it’s intercoms and paging systems.

Exploring solutions
With that type of solution in mind, we set out to find the best intercom system that would work with OnSIP. We tested three companies who have a hand in shaping this field and here’s what we found.

The field for competition in this segment of VoIP isn’t that large. When consulting with suppliers, customers and others who work in the VoIP industry, we kept hearing the same three names - CyberData, Digital Acoustics and Valcom. Each of these were well regarded by the market, so we wanted to put them head to head to find out which of these would be the best for our users.

We had problems with both the Valcom and the Digital Acoustics devices, unfortunately. The issues don’t discount their value within the field of SIP paging, but just mark them as being incompatible with OnSIP at this time.

Testing Cyberdata's offerings
The two Cyberdata systems which we looked at both excelled with the OnSIP system. Once we got them up and running, they were good, stable devices with excellent build quality. They were very easy to use and very easy to set up.

We tested the SIP-enabled IP Ceiling Speaker v2.0 and the SIP-enabled IP Outdoor Intercom. The most difficult aspect was finding the IP address of the devices in order to log into their webpages to set them up. We plugged in the devices onto our network and then checked the DHCP lease against the manufacturer’s MAC address. From there, we logged into the device and manually set a static IP address; this way, we would always know the IP address of the device, making maintenance easier.

We also found that the Nightringer functioned as needed for both devices. Nightringer is an option where, if a call is sent to the device, the device will broadcast a ringtone to let somene know that there is a phone call. This is typically handy in a warehouse situation. A speaker can broadcast a ringing sound to alert a late night worker that there is an inbound call. Cyberdata accomplishes this by creating a secondary user agent within their devices which does not broadcast speech, but rather upon receipt of the call will broadcast the ringing sound. For our testing purposes, we created a group with the Nightringer user and a Bria softphone, and we were able to replicate the expected behavior.

In our survey of the market for various paging systems, we learned that the three lead manufacturers all offer a fairly solid product. However, software issues hampered both the DigitalAcoustics and Valcom lines, making Cyberdata the line we found to work best with OnSIP right out of the box. Cyberdata was the only device to make it to the Junction Networks compatibility guide.

Are you going to the ChannelPartners Show? What type of reseller are you?

sukanya — Thu, 26 Jan 2012 22:14:27 +0000

By Channel Manager Andy Ogg

Are you an IT solutions provider looking for a reliable voice solution to provide to your clients? Are you a telecom interconnect specialist looking for a Hosted Voice Solution to compete in your market? Or, perhaps you are a business consultant working in a specific vertical, and you need a voice solution that provides features unique to that vertical.

There are many different types of business that resell the OnSIP solution. Leading up to the Channel Partners Expo we will be profiling different types of resellers. We will describe their businesses in general and how they became successful with OnSIP.

Today, I will be focusing on a business consultancy in Texas. They primarily focus on two verticals - education and medical. Analyzing operations costs and minimizing them is their value proposition. OnSIP’s unique pricing is perfect for these types of originations. OnSIP allows an unlimited amount of extensions with no extra charge. Doctors offices have exam rooms where they need phones that have low usage. And schools have low usage phones in classrooms. In the past 14 months, this business consultancy has deployed OnSIP at 10 organizations ranging in size from 7 to 100. They are currently earning about $1,200 in monthly recurring revenue from these deployments, and have earned over $9,000 in one time fees.

The OnSIP pricing model - paying for what you use and no more - allow these types of originations to greatly reduce their operating costs. In some cases, their telephone bills were reduced by up to 50%. Read our white paper on how OnSIP meets and exceeds the requirements of educational institutions.

A case study in this paper demonstrates how a typical school district with four geographically-distributed schools and about 230 users (270 faculty and staff, 2000 students) was able to reduce their phone bills by 50% while upgrading to a reliable VoIP system with more features. Such deployments enable agents to derive reliable, recurring income. Larger deployments in bigger school districts cost up to $10 million, thus yielding more returns.

Does any of this sound familiar to you? Are you interested in becoming an OnSIP agent? Are you going to the Channel Partners Expo in Las Vages in March? Please set up a time to come and talk to us.

Join us this Spring at Click here to make an exclusive appointment to talk about becoming an agent with our CEO.

“Now in HD!” Voice Codecs and what they really mean

nicole — Tue, 24 Jan 2012 20:26:12 +0000

This blog is by Eric Tamme, member of the OnSIP Engineering Team

What is a codec?

Codec is short for “coder-decoder”. A codec is a program, or an implementation of a algorithm, which converts an analog input signal, like your voice, to a digital signal by sampling it according to the algorithm. The codec can also take a sampled stream from a matching codec, and decode the samples back to a similar analog stream that was input originally.

Here is an example codec called PCM:

Now for the technical description – be warned, this is going to get dorky.

PCM, also known as G.711, is one of the oldest, and most basic fixed rate codecs. It is fixed rate because it samples the analog stream 8000 times per second. There is also a fixed range of the value a sample can be -127 to +127 to be exact, which is 256 possible values. This range is not coincidental; because of the way computers represent numbers internally, a value with a range of 0-255 (256 total values) fits perfectly into 8 bits. A bit is just a single binary value of 1 or 0, and in the base two system, it takes 8 bits to represent the values 0-255 (256 total values). So, the short of that is one sample takes 8 bits, and if we sample 8000 per second, our rate of sample is 64 Kilobits per second, or 64 Kbps.

What does this mean?

The basic idea is that the more samples you take, the smoother you will be able to replicate the original signal, and thus more effectively recreate the speech to the listening party. While this is for the most part true, there are some very clever people out there, and some very big research companies, who spend a lot of time trying to maximize bandwidth by creating more efficient codecs. That is to say, engineers are working on finding ways to digitally encode voice with fewer samples and/or lower the ranges of sample values, while still being able to recreate an audio stream from the digitized sample that is high quality. To give you some idea, here is a “short” list of common voice codecs:

AMR Codec
BroadVoice Codec 16Kbps narrowband, and 32Kbps wideband
DoD CELP - 4.8 Kbps
GIPS Family - 13.3 Kbps and up
GSM - 13 Kbps (full rate), 20ms frame size
iLBC - 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
ITU G.711 - 64 Kbps, sample-based
ITU G.722 - 48/56/64 Kbps ADPCM 7Khz audio bandwidth
ITU G.722.1 - 24/32 Kbps 7Khz audio bandwidth (based on Polycom's SIREN codec)
ITU G.722.1C - 32 Kbps, a Polycom extension, 14Khz audio bandwidth
ITU G.722.2 - 6.6Kbps to 23.85Kbps. Also known as AMR-WB. CELP 7Khz audio bandwidth
ITU G.723.1 - 5.3/6.3 Kbps, 30ms frame size
ITU G.726 - 16/24/32/40 Kbps
ITU G.728 - 16 Kbps
ITU G.729 - 8 Kbps, 10ms frame size
LPC10 - 2.5 Kbps
Speex - 2.15 to 44.2 Kbps
SILK - from Skype 8, 12, 16 or 24 kHz and a bit rate from 6 to 40 kbit/s

The most commonly used for voice are: G.711, G.722, G.729 and AMR in cellular phones.

Pass-through and Transcode

As mentioned before, some people spend a lot of time and money researching how to more efficiently encode analog streams and still achieve high quality playback. As a result, many codecs are not free, but must be licensed to be able to encode, or decode them. When you decode one codec format, and encode it to another codec format, it is called transcoding.

Pass-through means that there is no transcoding required the actual encoded bits are simply passed through an intermediary. For example, two endpoints (phones) support the DoD CELP codec, but the server they communicate through can not transcode DoD CELP. In this case the digitized signal will just pass through the server to the endpoints, each of which know how to encode and decode DoD CELP.

Will the real HD please stand up?

HD, in terms of voice codecs, is subjective because of the complex algorithms that are used to very specifically encode and decode the human voice. Some codecs use adaptive processing mechanisms to give more bits to audio ranges that are more discernible by the human ear by clipping the upper and lower ranges; this is called companding. Others change the sampling rate dynamically on a variety of factors. A good comparative argument is the quality of a CD versus MP3 encoded (another codec) sound. CDs are actually encoded using PCM, but sampled at 44,100 per second at 16bits per sample, which equates to 705.6 kbps per channel, while an MP3 could be encoded at 128 kpbs and have a reproducible sound that's nearly identical to many people. If you have an audiophile setup in your living room and a rack for hundreds, or thousands, of CDs - great! You have the ability to really tell the difference in audio quality between an MP3 and a CD. If you are running on a treadmill in a gym with a pair of crummy headphones, you probably can't tell the difference.

Best low bandwidth / mobile codecs

This is somewhat analogous to a network environment. If you are on 3G using a softphone on your smartphone, a wideband (higher kbps) HD audio codec will not sound any better; it will probably sound horrible if you can hear anything at all. A 3G data connection does not have the bandwidth to support a high kbps codec. Your best choice would be G.729, AMR, or GSM in a pinch. I list them in that order because, really AMR is a better low bandwidth codec than G.729, but virtually all SIP based phones you can buy will have G.729 support. Few will have AMR which means your intermediary would have to provide transcoding support, and that is not likely as AMR is almost always a pass-through codec.

Best high bandwidth codecs

If you are sitting at your desk and have a dedicated 10mbps hard wired circuit for your office, G.722 will give you excellent sound quality – really nearing the point of diminishing returns from a bandwidth to audio perception perspective. G.722 is an adaptive form of PCM, so it takes the same kbps, but is much more efficient at encoding voice. There are other codecs that may consume less bandwidth, but they are still high enough kbps rate that if you are experiencing issues, you might want to switch to a low bandwidth codec, or check your network for latency and packet loss problems.

Other considerations

You do need to think about compatibility when considering which codec to use. Both endpoints (phones) must support the codec, or the intermediate server between you must support transcoding for both your codec and the callee's.

OnSIP reviews the Konftel 300IP Conference Phone

leo — Tue, 24 Jan 2012 16:45:49 +0000

A customer recently asked us if we could recommend to him something other than a Polycom for his conference room. I realized then that we hadn't really looked into conferencing solutions from other manufacturers, and the team immediately sought out to remedy that.

We started by digging into the offerings of recently acquired Swedish telepresence provider Konftel. According to the company, Konftel's conferencing solutions capture 75% to 80% of the market in their home country of Sweden. Now the focus is on increasing their presence in the US and the rest of Europe.

The 300IP is the company's only SIP based solution. This rather odd looking device is packed with features. Highlights include built-in call recording, 5-way calls, preset conference 'groups', and OmniSound 2.0, Konftel's own version of 'HD Voice'.

Check out our full Konftel 300IP review.

Customer Feature: Brite Electric

sukanya — Fri, 20 Jan 2012 17:22:53 +0000

In the first customer feature of the year, we feature Brite Electric, an OnSIP customer for a year and a half. The family-owned company, founded in 1979, specializes in electrical and air-conditioning contract work and services the Central Florida area. Judy Wolek is the president, and she spent some time telling us what she likes most about OnSIP.

What phone service did you previously use and why you decided to look for other options?

At Brite, we have tried numerous phone services since the company was created. Initially, we had traditional POTS with the local telephone company. We then tried remote carriers and a number of companies over time. We had telephone service through Cablevision for several years.

One day, lightening hit our phone system and we were faced with having to replace it at a cost of at least $10,000. Then our IT consultant, who manages our computers, suggested that we look into VoIP as an alternative. That is when we came across OnSIP. We chose OnSIP over its VoIP competitors for its attractive pricing packages, which lent itself well for use in our office.

Tell us about your transition to VoIP, specifically OnSIP.

When we decided to try VoIP, we didn’t switch all at once. Instead of buying the equipment for a full phone system like our previous one, we ordered individual phone lines for all the key personnel in our office. All the field employees used their cell phones. We then bought one number registered with OnSIP and tried it. The calls were clear with no echoes and we were pleased with the quality. Then we started using another phone with OnSIP. We found that calls were uniformly of good quality. So we decided to go all out and use OnSIP fully for Brite Electric.

What are your favorite features?

My favorite feature while managing OnSIP is how the admin portal is online. I don’t have to be in the building to make changes. If the front office people are on leave, I can manage the system from anywhere, even after office hours. As an example, if I forget to change the greeting and office hours for a holiday, I can still do that when I remember, sitting at home. Very nice!

I have to add that, the first time I tried exploring the admin portal, I was a bit hesitant. I spent about 20 min on the phone with Larry Browne (member of the support team) who explained everything and set me off in the right direction. Even now, when I have any questions, I just have to call Larry and he helps me out patiently.

I’m a heavy speed dial user – I have more than a hundred customer numbers set to three-digit dialing and I constantly use the feature. I am a fan of voicemail-to-email, which tells me about important customers who need our help right away when I’m away from my office. I also have set up the system so as to receive texts when there are missed calls, which allows me to call back important customers when the front desk is overloaded.

How has your experience with OnSIP been so far?

I love that OnSIP is very reliable. In the year and a half that we’ve been OnSIP customers, we’ve had one minor hiccup and even that was solved immediately. I cannot say the same for the other phone systems we’ve tried.

And last, but not least, our savings using OnSIP is tremendous. With my old phone system, our bills used to be anywhere from $400 to $600. With OnSIP, our bills are never more than $150 for the same call volume and features. I could use the difference to pay for my truck!

We thank Judy for taking the time to share her experiences with us at OnSIP.

Counterpath and Vidtel announce interoperability: Multi-party video conferencing using your iPhone

leo — Thu, 19 Jan 2012 20:22:53 +0000

Today, Counterpath and Vidtel announced interoperability between the Vidtel video conferencing service and Bria ‘enabled’ computers and iPhones. Press release found here.

If you’re unclear about what Counterpath and Vidtel do, then that sentence probably left you scratching your head. Don't worry, I’ll explain. Counterpath makes software phones for your PC/Mac, iPhone and Android devices. Their leading software phone is Bria, and the company just recently released video calling features for the iPhone version. Now you might be asking, how does this differ from Skype, Facetime, or any of the other video calling services that are already available in the App Store?

Good question. With other video calling services, you’re limited to strictly that service or application. Facetime only works between Apple devices, and Skype users can only Skype with other Skype users, etc. This is not the case with Bria because it was built on open standards. Bria works with any SIP service provider (“network agnostic”), and its video calling feature should theoretically work with any SIP-based video phone (“device agnostic”). This means that I’m able to conduct a video call using my Polycom VVX 1500 with someone walking down the street using Bria on his or her iPhone. This is fantastic for video calls between two people, but what if I want to hold a video conference with multiple parties?

That’s where Vidtel comes in. Vidtel is an Internet video service provider that offers a network and device agnostic video conferencing service.

From the press release:

Vidtel’s MeetMe is a cloud-based service that enables any-to-any video conferencing between room-based systems, executive desktop video systems, PCs/Macs, smartphones and tablets using any combination of SIP, H.323, and other technologies.

Vidtel customers get a number or SIP address for a video conference bridge that they can dial into with any number of video phones. You can either pay for the video conference bridge per month (prices vary depending on maximum number of participants and whether you want HD or standard quality video) or simply pay per minute.

A real selling point for us is that you can use the bridge with people who aren’t even Vidtel customers. This includes people using SIP service providers. For example, OnSIP. We appreciate that Vidtel even went the extra mile and included interoperability with Google Talk.

We tested out MeetMe using video phones registered on the OnSIP network, and it worked flawlessly. Here’s an iPhone screencap of a MeetMe conference call between three people: one using Bria for iPhone, one using Bria for Mac, and one using a Polycom VVX 1500. Please excuse the "network quality issues" warning, as that is just a reflection of our questionable Internet in the New York City office.

And you know what the really cool thing about all this is? None of us had to do anything to get this working. As a service provider, we didn’t have to jump through hoops to make sure our OnSIP was compatible. As phone developers and manufacturers, Counterpath didn’t have to tweak Bria, and Polycom didn’t have to change any firmware on their VVX. Interoperability was there from the get-go.

I guess that just goes to show you that when companies build things adhering to open standards, really neat things can and will happen.

Top Reasons to Attend the Channel Partners Conference and Expo in March

sukanya — Tue, 17 Jan 2012 21:18:48 +0000

Recently, the Center for Exhibition Industry Research announced that the trade show industry was growing, with the number of trade show attendees jumping 3.7% in the third quarter of 2010 and the number of exhibitors by 3.6%, according to the CEIR index. With increasing attendance and sponsorships at such events, the product choices presented to participants will be diverse and numerous.

Trade show attendees, in general, say that attending shows helps them decide on what products to buy and that such shows impact their buying decisions because the competition is in one place allowing for comparison shopping in real time. Many say that trade shows help them become aware of new products and services. (source: AEM)

The Channel Partners Conference and Expo is, according to their website, the largest event for indirect sales organizations – agents, VARs, systems integrators, interconnects and consultants. The Spring 2011 event in Las Vegas drew nearly 3300 participants and almost 150 exhibitors.

We can think of many reasons to attend the Expo in Las Vegas from Mar 27-29 if you are an agent. Here are the top five:

1. Network with other resellers to discuss revenue generating ideas.

Learn what products and strategies your peers recommend. A Selling Techniques Luncheon will provide an opportunity to discuss success strategies with your peers. An Agent-VAR Partnering Workshop will help further your insight into the slew of solutions available today.

2. Keep up with the times.

Learn about the latest trends in information and communications technology along with successful implementation ideas from other agents. Pre-conference workshops and sessions on cloud services, mobile wireless, business development and new opportunities provide for great educational experiences. Attend an Industry Address by Tiffani Bova, the vice president for research of indirect channel programs and sales strategies at Gartner, when she will report on where the market is and where it is going.

3. Get in-person demonstrations of IT products directly from the companies.

Meeting face-to-face with product and sales teams will help you evaluate the many choices available, all in one place. With the long list of exhibitors vying for your business, you're bound to find new companies that you'd like to associate yourself with. All in a day's work!

4. Have fun while developing your business.

Cocktail receptions and fun mixers galore! Start and end every day with receptions and mixers, where you can meet other attendees in social settings. We don't have to introduce Las Vegas to you!

5. Meet the OnSIP team.

Learn about OnSIP's award-winning reseller program. A five member team from OnSIP, including Mike Oeth, CEO, Rob Wolpov, President, Tim Cleves, Director of Sales, Nicole Hayward, Director of Marketing, and Andy Ogg, Channel Manager, will be present at the OnSIP booth to explain how you can increase your revenues with OnSIP. Set up an exclusive appointment with Mike and Andy to discuss how OnSIP can meet your customers' and your needs.

Join us this Spring at Click here to make an exclusive appointment to talk about becoming an agent with our CEO.