VoIP Supply - ContactCenterWorld.com Blog Page 11
Make no mistake, almost everything is becoming a cloud based service. Still running Exchange? You’re living in the past, my friend. Phone systems are, of course, no different. While I’ll maintain there are huge advantages to running an on-prem system (mostly cost and low latency), there are a lot of conveniences of having your system in the cloud. Now, when I say cloud, I am referring to platforms like Microsoft Azure and Amazon Web Services for this specific post.
Let’s talk a little about the conveniences of a cloud hosted phone system. First, it makes deploying remote phones a much easier process, mostly because every phone is now remote. It also allows anyone traveling abroad to bring their phone with them, and with Internet access, they can make calls from Singapore as if they were calling from Buffalo, New York (for example) with no international toll charges. Of course, you can always call extension to extension for zero cost. That’s a pretty amazing concept.
You might be thinking, this can all be done with an on-prem system as well, and you’d be right, but why poke holes in your corporate firewall, and subject yourself to the fun of NAT traversal if you don’t need to? You can also accomplish redundancy with an on-premise system, but you will lack the flexibility of providing multi-region connectivity and redundancy (because it’s in the same building), which is what the above-mentioned cloud services can provide.
Why is multi-region connectivity important? Well, if you’ve been reading the news lately, you’ve probably heard that Amazon dropped an entire region for a couple of hours causing mass panic, and the zombie apocalypse (not really). This is the risk you take in exchange for convenience when you place an application or service in the cloud, but when you distribute that application or service across multiple regions, you mitigate that risk significantly. Some businesses went down entirely because they stuck all of their eggs into one basket (region).
It should be known that regions in these cloud services are treated as completely siloed entities. Instances in one region, cannot simply ping an instance in another region via local IP address, even if they are on the same Amazon, or Azure account. For that, you need some sort of connector, like a VPN. Be aware, however, this is accomplished differently based on what service you are using.
Amazon Web Services, for example, does not have any built-in tools at this time to connect regions together. If you’re planning on deploying FreePBX in both Oregon, and Virginia for redundancy, you’ll need to create a VPN between the two systems with your own virtual appliance so that they can exchange configurations. This should not be confused with Sangoma’s High Availability module for FreePBX, as that requires two systems to be on the same subnet with very low latency between them.
Microsoft Azure, DOES provide the ability to create a region-to-region VPN without using a 3rd party VPN concentrator, and with my experience, the more natively supported tools and services you use, the better things work overall. Truthfully, a VPN may not always be necessary, but that will be dependent on the specific phone system, and how it prefers to communicate with its slave or warm spare. It generally isn’t a bad thing to have regardless.
Before I get more into the strategy of multi-region redundancy, I’d be remiss not to mention a second option, which is connecting either Microsoft Azure, or Amazon Web Services to your local corporate network. Both services, have native tools to create a VPN to your network, provided you have a compatible firewall on your side of the equation. In this scenario, you would have a system on your local network, with a warm spare in the cloud, which can talk local IP to local IP. This option isn’t as flexible as moving all phone system communications to the cloud, but would still provide redundancy in the event your on-prem system goes down, but you still have a live Internet connection to your building. If your entire network takes a nose dive, you are SOL.
Strategy: I originally had the idea (when writing this post) of testing Wazo’s built in high availability module, but I found that just installing the platform on Amazon was so incredibly difficult and an inconsistent process that I just gave up. Back when it was called Xivo, I tested high availability and it worked great. It didn’t work as well as Sangoma’s High Availability module, but it did a decent enough job. The way that it works (or worked), is by moving the configuration from the master system to the slave via a secure tunnel, then it would synchronize and shut down Asterisk on the slave. Its job would then be to continuously ping the master, and in the event, the master was unresponsive, start Asterisk, and bring up the SIP trunks. The only thing you’d have to worry about is registering all of your phones to the slave PBX. That can be automated by using IP phones with a secondary SIP server.
>So, because Wazo was such a PITA, I decided to go with something more mature in the open source space for this post, FreePBX. FreePBX can be configured as a warm spare similarly to Wazo, but it isn’t as automated of a process. Take a look here, to see what’s involved in the basic setup. You will STILL employ IP phones with a secondary SIP server (>Sangoma’s phones do this BTW). Like WAZO, a transfer of the configuration is sent to the warm spare in the opposite region via a secure tunnel, but the difference is in the synchronization. Wazo will instantly synchronize, but FreePBX will require a restore to be performed, which can be automated. You will also need to exclude changing the network settings on the warm spare. We aren’t exactly replacing the production system, we are just providing an alternate for phones to register to. The only intervention that should be required in the event of a failover is activating the SIP trunks (because you would have chosen to turn them off in the warm spare’s restore).
To summarize: When your production phone system has an issue and goes down, your IP phones will attempt to register to the secondary SIP server (via public IP address), which resides in another region (using either Azure, or Amazon). To complete the failover, you will need to log into the warm spare, which has now become the production system, and enable the SIP trunks. Within a reasonably quick period of time, calls in and out will occur as if nothing happened.
While this all works, the primary challenge is the timing of the synchronization between systems since it is not instantaneous. Logically, you’ll want to back and restore to the warm space nightly, but if a lot of changes are expected on a system daily, you may want to schedule that more frequently.
If you plan on deploying your phone system to the cloud, and redundancy is going to be an important priority, well then, I hope I gave you something to think about. Stay tuned for my upcoming post on creating a quick and easy VPN between Amazon Web Services regions.
Publish Date: March 28, 2017 5:00 AM
VoIP News! The Latest on Several New Releases from Grandstream, Gearing Up for Channel Partner 2017 and more!
Things have been pretty busy over at Grandstream HQ as they have recently released a few new products! Grandstream has expanded their large VoIP catalog once again with the release of the GDS3710 IP Video Door System. This is an easy to manage surveillance solution that integrates with your IP communications and features a powerful 1080p video resolution!
Grandstream also released the GWN 7000 Enterprise Multi-WAN VPN Router is a features-rich powerful tool for any business and can be shared across one or many different physical locations!
Another new product released by Grandstream is the GWN7600 WiFi Access Point. This is a mid-level access point for small sized businesses or multiple floor offices and providers dual-band network throughput and expanded WiFi coverage range.
All of these products are available to order now!
As April is rapidly approaching, VoIP Supply is getting ready to exhibit at Channel Partners 2017 at the Mandalay Bay Convention Center in Mandalay Bay, Las Vegas! This will take place April 10th to the 13th and we will be focusing heavily on our Partner and Fulfillment Programs. If you are attending the show, please stop by booth 250 and say hello! We will be there to answer any questions you might have and we will also be giving away several Plantronics Headsets, so make sure you don’t miss out!And lastly, in VoIP News, VoIP Supply has hit a very special milestone as we are celebrating 15 years of business! It is an honor to work for VoIP Supply, and we have enjoyed solving problems and creating solutions for over 125,000 customers worldwide over the years. We are proud of everything the company has accomplished since 2002 and we look forward to the next 15 years!
Publish Date: March 24, 2017 5:00 AM
If you are in search of Polycom phones to update your current VoIP environment or to replenish a new office, don’t overlook our Refresh offerings. Our Refresh Polycom phones are great quality used phones that have undergone a 10-step reconditioning process to assure that you get the best of the best. And the best part of it all is that they come with a full six-month worry-free guarantee.
Watch below for the process of Examining, Upgrading, Setting to default, and Repackaging Refresh products.
Polycom VVX 310 (Refresh)
The VVX 310 is a Gigabit six line phone that provides crystal clear communications and enhances the workplace with collaboration tools and personal productivity features. Receive meeting reminders and alerts, manage your Microsoft Exchange calendar and see your colleagues’ Instant Messaging presence/status right on the screen of the VVX 310.
Learn more about the VVX 310
Polycom VVX 410 (Refresh)
The VVX 410 is a 12-line Gigabit IP Phone featuring High Definition (HD) Voice and a 3.5 color LCD display. This is a great phone for office environments handling a moderate volume of calls and also suitable for Unified Communications (UC) applications.
Learn more about the VVX 410
Polycom IP 7000 (Refresh)
A conference phone for all conference rooms, the IP 7000 provides high-fidelity conference calls for clear, life-life communication. With flexible configuration options and strong interoperability, the IP 7000 is a great choice not just for conference rooms, but for executive offices, boardrooms, and huddle rooms alike.
Learn more about the IP 7000
These are just a few of the Polycom Refresh options that VoIP Supply has to offer. Take a look at our Refresh Catalog to view more. With so many options at a great price and with a 6-month guarantee, we think you would agree when we say, “Why not buy Refresh?”
Publish Date: March 21, 2017 5:00 AM
Having trouble finding an ideal IP paging for your workplace? We’ll help you narrow down your choices by highlighting some of our top-selling IP paging products! Each of them has unique features that can fit in different work environments to ensure that you have an excellent communication experience. Let’s get started!
#1 Advanced Network Devices IPSCM
Ideal for: A workplace where it is difficult to install additional power for IP speakers, such as school baseball or soccer fields or outdoor campgrounds.
If you are looking for an easy-to-install IP speaker, you won’t want to miss Advanced Network Devices IPSCM. This solution is powered using an RJ45 connection on a CAT 5 cable from a PoE switch, so there’s no need to install power to function the speaker. It also required no special hardware or servers, making it so easy and quick to install.
The IPSCM is an 8in IP speaker embedded into a white 2 foot by 2 foot frame which is designed to be dropped in place of a standard ceiling tile in a typical drop ceiling.
“The IPSCM is easy to install since it only requires a CAT5 cable to connect to the network. It is also PoE capable so it can be powered through a standard PoE switch. Bonus awesomeness is that it’s a talk-back speaker for 2 way communications.”- Joseph Shanahan, Senior VoIP Consultant at VoIP Supply.
The Advanced Network Devices IPSCM Features:
- 8” Speaker
- Full multicast and broadcast support
- Field upgradable
- On-board web server for status and control and field upgrades
- Easy to install - built-in test tones allow quick checkout of installation
#2 The Algo 8180 SIP Audi Alerter (NEW)
Ideal for: A noisy or large workplace such as warehouses, restaurants and machines ships where you tend to miss phone calls because you just can’t hear the ringtones.
The Algo 8180 is a SIP compliant PoE speaker designed for voice paging, loud ringing, and emergency notification. With its high-efficient amplifier and loudspeaker, the Algo 8180 is able to generate an 8 times louder tone than a normal telephone ringer (which means no more missed calls!). The SoundSure technology adjusts loud ring and paging volume to compensate for background ambient noise, protecting your hearing from overly loud volume.
“The Algo 8180 is one of the easiest setups you can find, extremely loud, and very reliable. Instead of adding an extension or “seat” to the noisy site, you can simply pair the Algo 8180 with the Algo 2506 detector which detects the audio from the headset jack and activates 8180 alerter directly without using SIP.” - Brian Hyrek, Senior VoIP consultant at VoIP Supply
The Algo 8180’s most useful features include:
- Dual purpose loud ringing and talkback voice paging
- SoundSure ambient noise compensation adjusts output for noise level
- Selectable alert tones
- PoE eliminates local power supply
Access the Algo 8180 datasheet here
#3 The CyberData 011186 (NEW)
Ideal for: An outdoor workplace that faces extreme weather conditions and possible vandalism. Ex: Nursing care facilities, schools and universities, day care facilities, retail establishments, and highway tolls.
The CyberData 011186 is a versatile, cost-efficient outdoor intercom that offers a weather-resistant option and dry relay contacts for optional auxiliary controls. This SIP-enabled solution is perfect for settings requiring two-way communication and secure access.
“The great thing about the CyberData 011186 is the durability and outdoor weather resistant capability. It can also be powered by a PoE switch which really helps with mounting and installation.” - Darren Hartman, Senior VoIP consultant at VoIP Supply
The CyberData 011186 V3 Outdoor Intercom Features:
- PoE 802.3af enabled (Power over Ethernet)
- SIP compliant
- Adaptive full-duplex voice operation
- Network web management
- Network adjustable speaker volume and microphone sensitivity
Access the CyberData 011186 datasheet here
#4 The Valcom V-1020C (NEW)
Ideal for: Any size offices, retails, educational institutions, etc. where you need a reliable IP speaker solution.
Consider the Valcom V-1020C if you are looking for a reliable solution that can produce exceptional sound quality and wide sound dispersion!
The Valcom V-1020C is a one-way, 8in amplified ceiling speaker with a removable volume control knob for voice paging and background music. The sturdy steel grille design increases its durability for a long term use.
“V-1020C is a very strong solution that you can rely on for a long time. It produces excellent sound quality and is perfect for music reproduction.”- Jon Garbin, Experienced VoIP Consultant at VoIP Supply
The Valcom V-1020C Features:
- Utilizes Standard CAT 3,5, or 6 cable
- Excellent voice and music reproduction
- Built-in accessible volume control per input
- Additional pop-on, pop-off knob included
- Wide sound dispersion
- Electrically, acoustically matched components
- Sturdy steel grille
Access the Valcom V-1020C datasheet here
Publish Date: March 16, 2017 5:00 AM
If you spend a great deal of time deploying VoIP systems or building IP networks for various customers, you quickly learn to appreciate things that just seem to work. What I mean by that is, there’s no crossing your fingers, or sacrificing a virgin to have a piece of equipment work the way it’s actually supposed to.
If you’ve been reading some of my posts or have seen the videos that I have been in, then all of you know that I am a bit of an ADTRAN fanboy. I like ADTRAN the same way that I like my 22-year-old Swiss Army KnifeⓇ. It’s reliable, well-made, fits in your pocket, and will always get the job done. With that, let me re-introduce you to the ADTRAN NetVanta 1531P. For my first introduction to the NetVanta 1531P, check out my unboxing video here. Also see the configuration video below:
A key aspect that I’d like to spotlight on this switch is ADTRAN’s “VoIP Setup Wizard” feature. Setting up the LAN for voice is challenging, because of the complexity involved with configuring various settings such as Quality of Service (QoS), V-LANs, Class-of-Service (CoS), uplink ports, among others. With ADTRAN’s VoIP Setup Wizard, the whole process is automated to just two-clicks and your LAN is setup for voice. This is ideal if you have a voice over IP phone system or are thinking of switching to VoIP, which allows for fast VoIP deployments. ADTRAN offers a number of “voice aware” features that simplify setup and management of your voice deployments.
The NetVanta 1531P won’t actually fit in your pocket (unless you still wear Jenco jeans) but it has a fantastic form factor that allows you to mount or set it just about anywhere in your organization. Yes, it is a small capacity switch with 12 ports, which means it’s meant for small deployments of VoIP phones or wireless access points, but I’ve found that it really shines as a test or provisioning bench switch. The 1531P can function as a DHCP server, which is not a feature readily found in the lesser “prosumer” class of switch in the same-ish price point (under $600). This is important because you can serve IP addresses to IP phones without the need for a dedicated piece of equipment such as a router or server. You can of course consequently serve DHCP options, such as option 66, to point phones to a specific TFTP server. A customer’s environment can easily be replicated by creating an additional VLAN with their specific IP address scheme, default gateway, and DNS information allowing you to configure both phones and even an IP PBX to be shipped pre-configured, which is the actual method used by VoIP Supply for provisioning services.
Okay, how about some more details? Well, it’s a small, lightweight, fanless, PoE switch, and it’s a layer 3 device. Actually, layer 3 light as described by ADTRAN, which means it can perform static routing (VLAN to VLAN) but is limited to how many static routes you can use (16). It cannot perform dynamic routing like the NetVanta 1544 for example, but there are some pretty big differences in the price point when making that comparison (by a few thousand). The NetVanta 1531P sports 8 PoE and PoE+ ports which produce up to 30 watts per port (PoE+) with a maximum PoE budget of 65watts. In terms of IP phones, it’s plenty to support 8 PoE powered phones as they aren’t as demanding as wireless access points, but do your research and make sure you are accounting for all devices needing power. There are an additional 4 ports on the 1531P, two are non-PoE Ethernet, and two SFP, all of which are 1 Gbps. There is a console port accessible via a DB-9 RS-232 port that connects you to a very familiar command line interface. If you have any command line experience with Cisco, then you’ll feel right at home with ADTRAN as the syntax is nearly identical. Of course, you can SSH for the same experience, but you are also provided a web-based interface which means you’ll almost never have to SSH or console into the switch if you are a little rusty on your command line witchcraft. For a complete list of the NetVanta 1531P check out the datasheet.
ADTRAN’s standard web interface that you find with all of their NetVanta and most TotalAccess products provides you with an efficient avenue to administer your switch. It isn’t overly garnished with tacky graphics or is so convoluted that it becomes a frustrating experience if you don’t already know the ins and outs. The web interface is, just right, and leaves out virtually nothing. There are some tasks that are moderately quicker via the command line, but as stated before, the user interface can accomplish all of the most common switch administrative tasks without needing to use the command line. It also doesn’t change its appearance, ever. Changes or added features are subtle and go almost unnoticed if you didn’t read the latest firmware release. This isn’t necessarily a bad thing, as I tend to place value on consistency and general standardization. Speaking of firmware, it’s free, for the life of the product. Yes, I said FREE. Firmware can be downloaded from ADTRAN after you create a login with them, which is also at no cost to you.
If you think you have a need for a small layer 3 switch that can provide PoE to IP phones or any other compatible device, give VoIP Supply a call. The ADTRAN NetVanta 1531P goes to the top of my recommendations list for battle proven, reliable, and feature-rich equipment.
Publish Date: March 14, 2017 5:00 AM
Buffalo, New York, March 14th, 2017 - VoIP Supply, a leading VoIP solutions provider in North America, marked its 15th year anniversary with significant accomplishments in the CloudSpan Marketplace, a single platform that provides various cloud services to meet customers’ business needs.
In the past year, restructuring the CloudSpan Marketplace and the sales team to focus on proactive growth of service sales was a challenge for VoIP Supply but both are continually showing significant improvements the past 9 months.
Today, VoIP Supply’s CloudSpan Marketplace has grown mature, positioning for exponential growth in the current year and foreseeable future. The improved internal systems and processes satisfy new and existing customers with greater customer intelligence, increased team efficiency and effectiveness and a more personalized experience.
This year, VoIP Supply shifts the internal software platforms to more modern SaaS platforms and is expecting to see continuous improvements in customer experience and growth in the CloudSpan Marketplace as well as customer adoption of hosted VoIP services.
“The past 15 years have been a wonderful journey and a memorable adventure. I am extremely proud of what we have all been able to accomplish, the success that we have created, and the hundreds of thousands of customer interactions and solutions that we have been a part of,” said Ben Sayers, CEO and Founder of VoIP Supply. “We’ve experienced many great highs and have persevered through some of the lowest of the low. We’ve done so as a team and a family, working side by side with many great people over the years. I am thankful to all that have been a part of it and I’m grateful for all that we have been through.”
About VoIP Supply
VoIP Supply, LLC (http://www.voipsupply.com) is your trusted source for everything VoIP; from our large selection of name-brand hardware to our CloudSpan Marketplace. VoIP Supply provides you with a fully staffed inbound call center with licensed, certified and highly trained VoIP experts that can help you with any problem you might have. Whether you are a home user, business, reseller or service provider, VoIP Supply has the products, experience and expertise to make your deployment a success.
VoIP Supply is a three-time Inc. 500/5000 honoree, listed by Business First as one of WNY’s Most Admired Companies, as well as being consistently ranked one of Western New York’s Best Places to Work. VoIP Supply is also the first Certified B Corporation in Western New York.
Our dedicated Solution Specialists are here to help, so call us today at 1-800-398-VoIP or visit our website at http://www.voipsupply.com
Publish Date: March 14, 2017 5:00 AM
As school supplies, textbooks, technology equipment, and other related expenses are getting more expensive, many educational institutions are looking for a way to reduce their costs - switching to a VoIP system can help you achieve that goal. While a lower monthly bill is the most common reason for deploying a VoIP system, there’s a lot of other benefits you may not have noticed yet. Therefore, today we are going to walk you through some of the popular VoIP service benefits that education institutions love!
#1 Minimum Maintenance
With a hosted VoIP system, you don’t need an IT staff to maintain your phone system. Your service provider will be responsible for the system maintenance, monitoring, and management. Simple house your phones and the VoIP service provider will deliver your VoIP service through the Internet connection.
#2 Easy to Set Up
A hosted VoIP system requires minimal VoIP hardware and low upfront costs. Therefore, it’s easy to set up and get started (usually takes a few hours). Setting up a phone system quickly is a big benefit for schools who can’t close down too long during semesters.
#3 Unified Communications System
You would love unified communications if you have students or faculty in a remote area. With a unified communication system, conducting a virtual seminar/class becomes an easy piece. Schools are able to connect students, faculty, parents from different channels (social media, apps, email, instant message, etc.) into one single system.
#4 Paging/ Intercom Support
Paging and intercom support help educational institutions communicate efficiently especially during an emergency event. The integration of intercom and your VoIP system allows you to make important announcements right from your phone device.
#5 Phone Features
A VoIP system has a lot of potentials. Here we list out some of the popular features most school systems take advantage of:
- Announcements - make announcements about school events, schedule changed, or an emergency.
- Conferencing - conduct faculty meetings over telephones.
- Auto Attendant - set automatic professional greetings and direct callers to the specific routing options.
- Call Recording - record all important calls with parents, students, or faculty.
- Call Park - place calls into specific parking locations such as classrooms or offices andpickup the call when you arrive the location.
- Call Queuing - organize different school departments into queues so that callers can be routed to the right place.
- Find me/ Follow me call routing - control how your calls are directed even when you are not at the school office.
- Learn more.
Is your educational organization using a VoIP system? What do you benefit from it? Share with us!
Publish Date: March 10, 2017 5:00 AM
The world is changing fast. When I was little, no one was using the term “smartphone”, “USB port”, or “touchscreen”. Maybe, in the next decade, we will no longer need a so-called “phone” and the new generation will have to go to a history museum to see a physical phone.
The world is dynamic, so is your business. It’s time to review your phone system to see if it can keep up with your competitors, partners, and, the most important of all, your customers. Here we put together the top 5 indicators your phone system is outdated.
Top 4 Warning Signs Your Phone System is Not Sufficient for Business
Indicator #1 Increasing Monthly Fee
An increased phone bill is the most obvious indication your phone system need a facelift. While your competitors are cutting down their phone bills, your high bills might be putting you at a disadvantage.
Indicator #2 Decreasing Customer Satisfaction
In this competitive world, customers are expecting fast and precise answers from you. If your phone system can’t keep up their needs (call quality, etc.), you might lose your customers to other competitors who can serve them more efficiently and clearly.
Indicator #3 Lower Work Flexibility
With the help of technology, employees are able to enjoy higher work flexibility than ever before. More and more people are working from home or in a virtual system. If your phone system is not able to support or manage remote workers, you might face losing valuable employees or decreased work ethic.
Indicator #4 Separated Multiple Communication Channels
If your system doesn’t have a unified communication system, you might find your employees using various private communication channels. Not only could they miss important messages, but waste a lot of time making sure everyone is on the same page.
Indicator #5 Difficult to Upscale or Downscale
Businesses are dynamic. It’s normal to upscale or downscale your business at times, therefore, you have to make sure the changing process is as simple as possible. If you find it costly or time-consuming to add/remove a few lines, the phone system is not sufficient for dynamic businesses.
Now you have briefly reviewed your phone system. Did you see any warning signs in your phone system? What other signs did you see your phone system is holding back your business? Share with us!
Publish Date: March 9, 2017 5:00 AM
Our tech support team at VoIP Supply offers great pre- and post-sales support plus provisioning, consultations, configuration, and installation help. We get a lot of VoIP hardware and software questions and would like to share the solutions with everyone.
In the previous Mom’s calling Q&A series, we have discussed: Is There a Report that Outlines Grandstream Call Logs? Today, we have more new real questions and answers from VoIP users just like you.
What’s the default password for Plantronics Headsets?
Q: I have a Plantronics ear piece which I love. But then I got a new phone to replace my old phone and I can’t get the two paired. When I try to link the two I am asked for a pin number. I do not remember using a pin number when I first set this device up. If I did, I don’t remember the pin number. How can I solve this problem?
A: The default password for Plantronics headsets should be “0000“. If you are pairing a wireless headset, you can follow Plantronics headset pairing guide here.
Come back for more VoIP questions and answers next time! If you have VoIP questions to ask us, please submit a technical support ticket or contact our VoIP experts today at 866 582-8591.
Publish Date: March 8, 2017 5:00 AM
Looking for the whole lists of IP phones from Grandstream? We’ve got you covered! Follow this article to learn all the features and functions of Grandstream’s IP phones and the best workplace environments for each of their models. You will find the three main series: The GXP1600 series basic IP phones, the GXP1700 series mid-range IP phones, and the GXP2100 high-end IP phones.
The GXP1600 Series Basic IP Phones
Designed for small businesses who don’t require advanced features, the GXP1600 series are simple, easy-to-use IP phones that satisfy your basic needs of VoIP. The GXP1600 series comes with 3-way conferencing and easy no touch provisioning with other Grandstream products.
The GXP1610/1615 is a single line VoIP phone with a 132×48 pixel LCD display and 3-way conferencing. With up to 2 call appearances while handling a single SIP account, the GXP1610/1615 offers high-end phone features that can be depended on in various office settings.
The GXP1620/1625 is a 2-line IP phone with a 132×48 pixel backlit graphical LCD display. Offering 3 XML programmable soft keys and multi-language support, this solution gives you the flexibility you desire.
If you are looking for a basic solution that can be interoperable with most third-party SIP device, the GXP1628 would be your go-to solution. It is a 2-line IP phone with 3 XML programmable keys and dual network ports that offers you HD wideband audio, full duplex speakerphone, and more features.
The GXP1630 is the latest member of the 1600 series that’s equipped with 8BLF keys and 4-way conferencing abilities. Integrated PoE, 3XML keys, dual switched Gigabit ports are just some of the useful features our customers love.
The GXP1700 Series Mid-Range IP Phones
The GXP1700 series features high-end, modern design, mid-range capacity at a near entry-level price. With up to 8 lines and 4 SIP accounts, this series offers 5-way conferencing, 4 soft keys and integrated PoE, USB port, and EHS support. The 24 to 32 digitally programmable keys and the advanced call productivity, efficiency, and high-volume control give users the flexibility they need to grow their businesses.
The GXP2100 Series High-End IP Phones
The GXP2100 series IP phones are tailored to businesses who are looking for more advanced features to maximize the work productivity and efficiency. Offering up to 12 lines, 6 SIP accounts, and 48 virtual BLF keys, the GXP2100 series highlights its suite of advanced call handling features that maximize your work functionality. See the chart below for a detailed comparison of this series:
Fill out the form below to download the full resource guide:
What IP phones are you using? Did you make the right choice? Share with us!
Publish Date: March 6, 2017 5:00 AM
VoIP security is a hot topic, and rightfully so. A compromised system can cost you $$$ in phone bills, so how do you prevent a breach? Well, the answer isn’t as complicated as you’d expect. There are a lot of opinions floating around on the subject, so let me address some truths and falsehoods that may be of importance when securing your VoIP system.
Fiction: You NEED a session border controller (SBC)
If you are a small business or are installing a VoIP system in your home, there is no need for an SBC. An SBC is a great device (or virtual appliance) because it masquerades your internal VoIP infrastructure. In basic terms, a SIP trunk from a provider terminates to the SBC, which then connects to your phone system via a SIP trunk. The SBC acts as the middleman in the transaction. To an outsider, SIP header information sources from the SBC and not your internal equipment. Although an SBC is a great extra layer of security and reduces overall attack vectors, it’s not required to make VoIP reliably secure for the majority of small deployments. Terminating a SIP trunk directly to your phone system behind a hardware-based or virtual firewall provides the security that would be deemed required to keep you incurring fraudulent toll charges.
Fact: You NEED a firewall
On the same topic as above, if you are going to be using SIP trunks to talk to the outside world, you’ll need a hardware or virtual firewall appliance to secure what is allowed in and out. In addition to the basics of protecting SSH, Telnet, and HTTP/HTTPS access to your phone system, you should always restrict what IP addresses can communicate directly to the phone system when it comes to SIP, and IAX (if you use it). What that means is only allowing IP addresses from your SIP provider, any remote extensions, or remote branches. Never ever expose your system directly to the internet without some type of firewall in front of it.
Fiction: Remote extensions MUST use a VPN
This is not true but isn’t a bad idea. A VPN will allow you to bypass NAT, which is the culprit in most one-way audio issues. The trick here is to tell the phone system all of the local IP subnets that it will be talking SIP. You’ll find this to be configurable on just about every Asterisk based phone system. A VPN also allows you to encrypt your session if you’re worried about the NSA listening in. An alternative would be using TLS and SRTP without a VPN, but you’ll just lose the benefit of avoiding NAT. The best way to securely deploy remote extensions is to use either a VPN or TLS. If you’re not using a VPN, make sure to define your inside IP subnets (as mentioned before), as well as your external IP address. These are all also configurable on just about any Asterisk system. Make sure you port forward SIP and RTP in your firewall to your phone system and secure your inbound rules by source IP addresses. Every system is a little different, but most Asterisk systems use 5060 UDP (SIP), and 10000-20000 UDP (RTP).
Fact: VoIP is NOT set it and forget it technology
If you’re going to take on the task of managing an IP phone system in your IT infrastructure, you need to adopt the mindset of monitoring it. Especially if you have port 5060 open to the outside world, you need to be logging and enabling alerts. In the past, phone systems have been bolted to a wall in a closet that no one ever went into except the PBX vendor. Now your system is racked next to your switches and servers. For those of you who are FreePBX users, Sangoma has just started to release their RMS platform, which simplifies centralized remote monitoring of multiple FreePBX and PBXAct systems. Stay tuned for a review on this!
Fiction: Not using port forwarding makes your phone system more secure
This isn’t actually a common belief, but it comes from a post I recently read on Spiceworks. It was claimed that a system has been made more secure by not forwarding port 5060 UDP from the firewall to the actual PBX. If this configuration was actually working, it was a minor miracle. The fact is there are usually two components of sending SIP traffic through your firewall. There is a firewall rule, allowing the traffic, and a fixed NAT association with the protocol and a device within your network. As long as you’ve made appropriate rules allowing SIP to your system, the port forwarding is simply a mechanism to help keep consistent NAT associations. In general, SIP and NAT do not play well with each other. Pro TIP: when you experience one-way audio, always look at NAT first.
Fact: You do not need to restrict RTP traffic to specific source IP addresses
I bet you never thought of this one. If you have, bonus points. While you should ALWAYS restrict SIP traffic by source IP address, it’s not necessary to do so with RTP. RTP is simply a media stream and doesn’t have the capability of initiating a SIP session, or any kind of session. Dare I say, you can leave the RTP port range open on your firewall. However, it doesn’t really hurt anything to place a source IP restriction on it.
Publish Date: February 28, 2017 5:00 AM
Are you a Skype for Business (SFB) user but not using SFB-optimized IP phones? Then, you are really missing out!
Using a SFB optimized VoIP phone can not only increase your call quality, but also make your work more effective and efficient. In this multi-media world, communication is more than just talking on the phone. Outlook integration, Instant Message, Voicemail and Videomail are just some of the useful features countless professionals are benefiting from using an optimized IP phone.
So, follow us to explore the best SFB-optimized IP phones that can give you a better communication experience!
The Polycom VVX600
The Polycom VVX600 is one of the most popular options for Skype for Business users. Built with executives and managers in mind, the VVX600 features a 4.3” gesture based, multi-touch cable capacitive touchscreen LCD display. This powerful solution comes with the capabilities to manage Microsoft Exchange Calendars, receive meeting reminders/ alters, access the corporate directory and Instant Messaging/presence status, etc. With the web-based, intuitive configuration tool, users can deploy and manage the system easily and quickly. Click here to download the Polycom VVX 600 Datasheet.
- 16 line registrations
- 4.3in LCD gesture based, multi-touch capable capacitive touchscreen
- Voicemail and video mail support
- Dual USB ports (2.0 compliant) for media and storage applications
- Dedicated RJ-9 headset port with EHS support
- Two-port Gigabit Ethernet switch
The Mitel MiVoice 6725
Choose the Mitel MiVoice 6725 if you are looking for a full featured IP phone optimized for use with Skype for Business. The MiVoice 6725, a certified Skype for Business compatible VoIP phone, offers full color 3.5” LCD screen and a separate Unified Communications (UC) presence icon for direct access to features and presence indications in Lync. Its exceptional voice quality allows you to hear and be heard clearly. Professionals love its easy integration with Lync destop, Outlook calendar, contact, click to dial and much more!
Popular Features and Functions of Mitel MiVoice 6725:
- Embedded Microsoft Lync 2010 Phone Edition Software
- Color Screen
- Dual Gigabit Ethernet ports
- Direct desktop integration
- Personal presence indicator
- USB interfaces
The Snom D725
The last but not the least is the Snom D725. Coming with 12 identities and 18 two-color multi-purpose LED keys, the Snom D725 is a flexible Skype for Business certified solution designed for busy multi-taskers just like you. If is a Gigabit VoIP phone that offers wideband HD audio, PoE, USB port and more. The D7 expansion module and the 18 programmable keys give users the highest flexibility to meet their business needs. Access to the Snom D725 datasheet. Watch the product video below to learn more!
The Snom D725 Features and Functions:
- 4-line B/W display
- 18 LED function keys
- 12 SIP identities
- Wideband audio
- Hands-free operation
- Power over Ethernet (PoE)
What kind of IP phone are you using for your Skype for Business? What’s your experience? Visit our website or contact one of our VoIP experts at 1-800-398-VoIP to learn more Skype for Business optimized IP phone options today.
Publish Date: February 27, 2017 5:00 AM
Our tech support team at VoIP Supply offers great pre- and post-sales support plus provisioning, consultations, configuration, and installation help. We get a lot of VoIP hardware and software questions and would like to share the solutions with everyone.
In the previous Mom’s calling Q&A series, we have discussed: What SIP Phone Could Connect Wirelessly? Today, we have more new real questions and answers from VoIP users just like you.
Is There a Report that Outlines Grandstream Call logs?
A: Is there a User Guide that outlines the various data fields the Grandstream captures on calls in and out? We are looking for good data that we can parse into a reporting package to manipulate/ report on that data.
Q: Basically, it is the API/CDR report, call log. You are able to capture the data and export a csv. file. However, for more advanced features you would need to utilize AGG software shown here:
- Here is the logger: http://www.aggsoft.com/pbx-data-logger.htm
Come back for more VoIP questions and answers next time! If you have VoIP questions to ask us, please submit a technical ticket or contact our VoIP experts today at (866) 582-8591.
Publish Date: February 22, 2017 5:00 AM
If you were expecting to read about ShoreTel, Cisco, or Avaya, you’d be wrong. While these are “VoIP” systems, I’d like to draw some attention to the new world order of IP telephony. It’s no secret that the big names in VoIP (mentioned above) are suffering, and it’s all thanks for a little piece of software called Asterisk. Not to discount the contributions of Freeswitch, but Asterisk is, go here for a little background.
Many have harnessed Asterisk to create some pretty great turn key solutions. Below are my top 3 recommended Asterisk systems from years of deployments and obsessive tinkering. Behold, my shortlist.
Grandstream UCM 6200:
Previously the 6100 series; this appliance based IP PBX is a light weight, small (desktop/shelf mount, and 1U rack mounted) appliance that offers a range of 30-100 concurrent calls, depending on the model. The fact that this is PoE powered, and has built-in FXS and FXO is huge. The interface is excellent, and is genuinely intuitive. This appliance demonstrates that VoIP isn’t exclusive to specialists in that field, and can be integrated into existing environments by network, and system admins alike. This has always been a huge favorite of mine, which largely has to do with the company that makes it. Grandstream is endlessly innovating, and improving their product lines. It requires very little maintenance, and is a reliable performer. I personally have never seen one have a physical component failure. For those of you who prefer to pay a one time capital expenditure, this might be a system you should strongly consider. With that said, it’s priced to sell! Check them out here.
It would be astoundingly foolish not to include FreePBX on my top 3 list. It’s probably the most recognizable name is regards to its ties to Asterisk and open source VoIP. What you may not realize is that it’s a fully matured enterprise ready PBX. It’s also one of the most versatile platforms that can be easily deployed on an appliance like VoIP Supply’s Renegade PBX, Sangoma’s own appliance, or virtualized on Hyper-V, or VMWare. FreePBX goes down in my book as the best “bang for the buck” in that there is no cost to have a fully functioning system. There are additional commercial modules that make life better, but aren’t required to make it work and there is no imposed limitations. Its integration with SIP Station also makes it an attractive solution. Filling out a short form in FreePBX’s web interface will have you calling out, and receiving calls in just a few minutes.
Digium excels at user interfaces, and also bells and whistles. The user interface is designed to be used by humans, and never really becomes overwhelming to non-VoIPers. Speaking of bells and whistles, Switchvox’s Switchboard is monumentally amazing, and useful. The Switchboard has been around for a while now, but is much improved with Switchvox version 6.X and is a feature that really sets this system apart. Seriously, check out the demo video. Generally, Switchvox makes life better in a call center environment and really hardnesses Asterisk’s potential. They should though, because they created it. Aside from the Switchboard, the biggest differentiator for Switchvox is how you build IVRs, or auto attendants. Instead of defining a destination for a keypress (as with most systems), you build sequenced actions and functions allowing you to create a very capable menu structure. Of course, Asterisk (on other platforms) gives you the ability to create infinitely configurable IVRs through scripting, but Switchvox gives you this functionality through its web interface.
Honorable Mentions: Keep an eye on these.
Ombutel is a derivative of Xorcom’s iteration of an Asterisk based platform, and it’s great. The user interface is clean, makes sense, and it supports FOP2 (flash operator panel). As with FreePBX, Ombutel is 100% free and virtualizes nicely. It is however still a little lacking in features when you compare it to FreePBX, and out of the box support for certificates is non-existent. I would like to see a self-signed certificate out of the gate at the least. Because of this, installations should be limited to on-prem. However, they are constantly adding features and fixing bugs. A major differentiator for Ombutel is their Class of Service (CoS) feature. This allows administrators to finely restrict or allow access to parts of the phone system from internal extensions. Normally, this is a features you’d have to pay for with other systems. Ombutel has a very active online community and are constantly receiving and replying to feedback. If Xorcom keeps this product alive, I expect it to do well.
Wazo, formally Xivo, is another 100% completely free system that is also feature-rich, but has a rare High Availability feature right from the get-go. High Availability is typically an expensive add-on, and that makes Wazo a really interesting product. Wazo is the most rough around the edges system in this list, but it has a lot of potential. Before it was Wazo, I found Xivo easy to install, but since they’ve gone through some changes, I haven’t been able to get Wazo to behave correctly when virtualized. I found it to be very buggy and dysfunctional. With that said, I am looking forward to the future development of Wazo. It’s very much a PBX un-plugged. If it were a car, it would be a driver’s car.
Publish Date: February 21, 2017 5:00 AM
Are you looking for an affordable way to update or get started on your VoIP solution? Then you are on the right page (literally!)
Refresh is our own line of Refurbished products that offer end users and businesses an affordable way to equip their offices or homes with VoIP equipment. VoIP Supply has a great array of Refresh products that have been examined for functionality and that underwent a 10-step reconditioning process before being repackaged. This month we are featuring three great products. Check them out below!
This Polycom favorite is a 6-line IP phone that is perfect for call center environments and/or secretaries and receptionists handling a low to moderate volume of calls. The VVX310 features HD Voice for lifelike conversations and an intuitive, easy to use interface. This phone also brings unified communications (UC) features and productivity tools.
For instance, the VVX310 complements workplace applications by allowing the users to view and manage their Microsoft Exchange Calendars, receive meeting reminders and alerts and access Instant Messaging/presence status right on their VVX phone display. Get the datasheet here.
Polycom IP 5000
The Polycom SoundStation conference phone turns ordinary conference calls into crystal-clear conversations that will have you thinking you are in the same room as the person on the other side of the call. This conference phone is perfect for executive offices or small conference rooms. Features of the IP 5000 include HD Voice, 7-foot microphone pickup and compatibility with a broad array of SIP call platforms. Get the datasheet here.
Snom 870 Black
This Snom phone is a great option for users looking flexibility and future-oriented technology.
The Snom 870 supports 12 lines and has a 4.3″ color TFT touchscreen display. But the best part of this phone is that it is Gigabit and it can support a WiFi card through the integrated USB port so that you can bring your Snom 870 wherever your wireless local area network (WLAN) can reach. Get the datasheet here.
Worry Free Guarantee: We stand behind our Refresh line and know that our products are quality products that will look great and work even better. But for your peace of mind, all Refresh products come with a 6-month guarantee, so that you can buy with confidence knowing that any mechanical failures and defects will be covered.
To learn more about Refresh or any of our product lines or services give our VoIP experts a call at 1-800-398-8647.
Publish Date: February 21, 2017 5:00 AM