Can you SMS through VoIP? Which VoIP Providers Support Texting Messages?

September 18, 2020 by Ying-Hui Chen

What is SMS? 

SMS, also called texts, stands for short message service. It’s a messaging service that can contain up to 160 characters. 

Why SMS? Some businesses rely on the SMS service for marketing campaigns and/or business notifications such as sales reminders, promotions/special pricing alerts or emergency alerts, important notification messages, and more. As more businesses are transitioning from traditional analog lines to VoIP, the demand for texting (SMS) through VoIP also increased. 

Can You Text (SMS) Through VoIP?

Great news – Yes, it can be done through VoIP! SMS through VOIP allows users to send texts through an Internet connection. In order words, you are able to send text messages to cellphones from your VoIP devices.

By texting/SMS through VoIP, your business can reach out to a broader customer base at once and save more money. While most VoIP service providers charge a fee for SMS messaging service, it’s generally much cheaper than a traditional cellphone plan. See below for a quick list of some of our popular service providers that support SMS:

Which VoIP Providers Support Texting/SMS Service?

VoIP Supply works with multiple VoIP service providers that support SMS texting service. We will work with you to find you the best solution for you. Here’s a quick look at some of our popular options:

1. Sangoma SIPStation

Sangoma SIPStation provides SIP Trunk service. Customers can sign up for SIPStation service with SMS-capable numbers and SMS metered service. 

Users also have the option to use their existing phone numbers for SMS. If you are using Switchvox, you can set up Switchvox with a SIPStation VoIP provider and a Switchvox SIP Phone extension to use an SMS DID. Learn more here.

2. Nextiva Desktop App

Nextiva support sending and receiving SMS service. Users need to activate Nextiva business numbers for texting (SMS), create a new contact for each SMS recipient, and format the phone number to Nextiva specifications. Learn more here.
3. Broadvoice B-Hive Communicator

Broadvoice empowers teamwork anywhere from any device. Its service allows users to send and receive SMS text messages to any number directly through B-Hive Communicator within your desktop. 

Broadvoice Communicator also includes a beta version of video calling! This feature is perfect for small team communication, allowing group chats to move to video with one simple click. The best part? No download is necessary! Learn more here.

See More VoIP Service Providers with SMS service here!

Not sure which provider is ideal for your business? Give one of our VoIP experts a call at 1-800-398-8647 today! We will be happy to give you a free consultation with multiple quotes!

Grandstream UCM6300 Series IP PBX Comparison

September 16, 2020 by Ying-Hui Chen

Grandstream’s UCM6300 series is a new series of high-end UCM IP PBXs. The UCM6300 series includes four models, the UCM6301, UCM6302, UCM6304, and UCM6308. Let’s check out what are the differences and which one is ideal for your business!

Download our webinar presentation slides here to learn more!

The Key Features of the UCM6300 Series:

  • A true enterprise-grade unified communications solution
  • Increased capacity – up to 5000 users 
  • Included video conferencing platform 
  • Wave mobile app for communicating with all endpoints 
  • UCM Remote Connect – available cloud SBC / NAT Traversal 
  • Built-in high-availability and hot standby 
  • API available for integrations and interoperability (Asterisk 16)

How do you choose the  UCM6300 IP PBX model that is right for you? It depends on the capacity you need. How many users do you have? How many concurrent calls do you usually handle? Any FXO/FXS needs? See the chart below to compare these four models side by side:

The UCM6300 Series IP PBX gives you up to 8 FXO/FXS ports. See the layout of each model here:

The UCM6300 series also comes with more advanced features such as:

Built-in conferencing platform

  • Built-in and fully included – no extra cost 
  • Supports desktop (WebRTC), mobile (Wave app), SIP video conferencing endpoints, IP phones 
  • 2, 3, 4 and 8 rooms 
  • 10, 20, 40 or 70 video participants 
  • 75, 150, 250 or 300 audio participants 
  • Video codecs: H. 265, H.264, H.263, H.263+, VP8

Wave Desktop – Wave Mobile App

  • For iOS and Android 
  • Mobile communications hub for the UCM6300 series 
  • Free to download and use 
  • Communicate with any device or user integrated with the UCM6300 series 
  • Integrate with UCM Remote Connect (Not with Wave Lite ed.)

UCM Remote Connect Service

  • Cloud NAT traversal service from Grandstream 
  • Create secure remote connections between remote users/devices and the UCM6300 series 
  • Ideal for any business with remote workers and/or multiple offices 
  • Create secure connections between remote UCMs 
  • Plans and specifications coming soon

Still not sure which UCM6300 model is the best for your communication needs? Call one of our VoIP experts at 1-800-398-8647 today for an evaluation! The UCM6300 series is coming soon! Stay tuned for more information.

VoIP Supply’s Refresh (Used) IP Phones for Less

September 15, 2020 by Youleidy Vega

We want to continue sharing VoIP Supply’s Refresh Program for refurbished devices because we know that there are so many businesses out there that may be in need of just that- used IP phones- but may not exactly know where they can get quality refurbished devices for fair market value. 

VoIP Supply has been in the refurbished game for years now, and we are not perfect, but we are pretty close to it! We are confident that we are hard to beat when it comes to offering used devices that work great and look good. 

Take a look at our 10-Step Refresh Process: 

These are our featured products of the month! 

The Polycom VVX 501 IP Phone is a Gigabit device with a touchscreen display. It provides all the key features a professional user would require in order to work effectively and efficiently. For example, the VVX 501 complements user’s desktop computers by displaying calendars and corporate directories right on the phone’s display. With Polycom’s HD Voice in both the handset and the speakerphone, rest assured that your calls will be crisp and clear.

Polycom VVX 501 Datasheet 

Another great refurbished IP phone, also from a respected VoIP manufacturer is the Yealink T46S. 

As you can see in the picture above, the T46S is a beautifully designed IP Phone with a color 4.3-inch color touchscreen display. It has a paperless design and supports up to 16 SIP accounts. The T46S supports Wi-Fi via the WF40 dongle, and Bluetooth via the Grandstream BT40 dongle. 

Some other top features include:

  • USB2.0
  • Opus
  • Headset and EHS support
  • USB recording
  • Support for expansion modules
  • Stand with two adjustable angles
  • Wall-mountable

Yealink T46S Datasheet 

And for our friends looking for a refurbished conference phone, the Polycom Trio 8800 is the way to go if you need a noise cancelling phone, with HD Audio that also has a familiar Skype for Business touch screen interface. 

With built-in Wi-Fi and a 20 foot range, the Trio 8800 is a reliable and flexible conference phone that can be in small-to large meetings. It supports 5-way conferencing and has Polycom’s patented noiseblock. 

Polycom Trio 8800 Datasheet 

These refurbished options make a great buy for users that want quality for less. These devices are backed by VoIP Supply’s one year warranty for your peace of mind! 

Have questions about VoIP Supply’s Refresh? Give us a call at 1-800-398-8647. 

Does SIP Trunking Save My Business Money?

Note: This article was originally written by Garrett Smith, and edited by Ying-Hui (Evy) Chen

Q: We recently purchased an IP PBX, but aren’t using VoIP to send and receive calls. I keep hearing about ‘SIP Trunking’ everywhere I turn. What is ‘SIP Trunking’ and can it help my business save money on calls?

A: If you are the proud owner of a SIP compatible IP PBX and are currently using a PSTN connection, such as a PRI or a handful of POTS lines to make and receive phone calls, it might be time to give SIP trunking a look. 

Many IP PBX systems, such as Asterisk and Trixbox allow you to leverage SIP trunking, in addition to maintaining PSTN connectivity. By editing the dial plan in your IP PBX, you can dictate which types of calls you want to connect to the PSTN, and which calls you want to hand off to your SIP trunking provider for termination. Cost savings is a primary reason why SIP trunking is becoming increasingly popular with businesses of all shapes and sizes.

What is a SIP Trunk?

For enterprises wanting to make full use of their installed IP-PBXs and not only communicate over IP within the enterprise, but also outside the enterprise, a SIP trunk provided by an Internet Telephony Service Provider to connect to the traditional PSTN network is the solution. 

Unlike in traditional telephony, where bundles of physical wires were once delivered from the service provider to a business, a SIP trunk allows a company to replace these traditional fixed PSTN lines with PSTN connectivity via a SIP trunking service provider on the Internet.

Download our SIP Trunking guide here to learn more!

What are the benefits of using SIP Trunking?

There are a number of reasonable answers to that question, but the primary advantage for many users is cost savings. For example, as a business owner, I could leverage SIP trunking to bypass expensive Long Distance or International toll charges levied by my PSTN provider. The rates for domestic and international LD are typically a fraction of what can be charged by traditional Telco operators. 

Understanding that a SIP trunk is dependent upon the integrity of your data (internet) connection, it is often prudent to use SIP trunks to complement traditional PSTN circuits. This gives you the cost savings you are looking for, in addition to the fail-over and redundancy of having two call carriers available.

More SIP Trunking benefits include more flexibility, DIDs, and more. See this post to learn more: 9 Benefits of SIP Trunking

How do I set up SIP Trunking on my IP PBX?

Actual trunk configuration will vary depending upon the make and model of IP PBX you are using. 

In a nutshell, you need to configure SIP trunking using account-specific information obtained when you sign up for service with an ITSP such as 

Next, modify the dial plan in your PBX to set up specific call rules which dictate which trunks (SIP or PSTN) calls are placed on when a user picks up a phone in your office and dials an outbound call. A very simple dial plan setup would use PSTN trunks for local calls, and SIP trunks for LD or International calls. Many businesses with multiple domestic and international locations use SIP trunking for intra-office calling as well, often leveraging their existing data infrastructure.

If you are interested in getting SIP trunking in place feel free to contact one of our VoIP experts at 1-800-398-8647 and our team will be happy to share some instruction for your specific PBX platform.

How to Connect Jabra’s Evolve 40 to a PC or a Mobile Device

September 11, 2020 by Ying-Hui Chen

Jabra headsets are very intuitive and easy to use. Today we will show you how to connect your Jabra Evolve 40 to your laptop or a mobile device. 

How to Connect Jabra’s Evolve 40 to a PC

The Jabra Evolve 40 headset connects to your PC through a controller. Fully plug your headset into the controller first and then connect the controller to a USB port on your PC. Make sure your headset jack is fully inserted into the controller.

What happens when the headset is disconnected from the controller during a call? The active call will be put on hold until you reconnect the headset to the controller and manually resume the call on your PC.

How to Connect Jabra’s Evolve 40 to a Mobile Device

When connecting your headset to a mobile device, you need to remove the controller. You should simply plug your headset directly into your mobile device. Note that when the headset is connected to a mobile device, the overall call quality will be lower than when connected to a PC or tablet through the controller. 

There you have it! Have you successfully connected your Jabra Evolve 40 to your device? Visit our product page to learn more!

Sangoma Reseller Updates – Switchvox Cloud, SIPStation, and FAXStation

September 9, 2020 by Ying-Hui Chen

This month’s Sangoma reseller webinar is loaded with good information! We took a closer look at Sangoma’s Switchvox Cloud, SIPStation and FAXStation. If you missed the webinar or simply want to refresh your memory, here’s a quick recap:

What’s New in Switchvox Cloud 7.3.1?

Switchvox 7.3.1 is the latest version of Switchvox, which brings several new features to Switchvox Cloud. Some key features include:

  • Emergency dialing notifications: When a user dials 911, a notification is sent to the email address(es) specified.
  • Music on hold
  • Easy call pickup
  • Switchvox Desktop software: The Switchvox Desktop Softphone for Windows and macOS is now available
  • Learn more here

Download our webinar slides here to learn more!

During the webinar, we also talked about the benefits and features of SIPStation and FAXStation. 

FAXStation solves problems such as:

  • Use fax machine and or efax
  • Low monthly cost- no per user charges
  • Will work in low bandwidth situations
  • Avoid T.38 compatibility issues
  • Learn more here

SIPStation features:

  • Trunk Groups: Sharing capacity amongst multiple locations
  • T.38 Faxing
  • Failover Options:
    • DID based
    • IP Address/FQDN
    • Global Failover
  • Bursting: Pay per min inbound/outbound when going above trunk capacity
  • Built in Fraud Guard: International calls are capped to prevent costly toll fraud
  • Learn more here!

Sangoma Reseller Webinar Q&A Session

Question: How can a Sangoma partner benefit from selling SIPStation or FAXStation? Do we have a different price list?

Answer: You don”t have a different price list. You’re able to get recurring revenue. When you sell through VoIP Supply, you get access to SPIFFS. Currently, there is a 5X MRR SPIFF. See the details here.

Have questions? Contact one of our VoIP experts at 1-800-398-8647 to learn more!

Is SIP Calling Free?

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. 

Q: Is SIP calling free? How much does SIP calling cost?

A: It is not free but it’s fairly cheaper than a traditional landline. Learn more here: Is VoIP Cheaper than a Landline?

  • One single provider for all: A traditional landline sends communications through an analog private branch exchange (PBX) system while a VoIP system transfers voice and data via a single network. There a VoIP system eliminates the need for multiple service providers. This will cut down your expenses significantly!
  • No long distance call fees: Since all VoIP phone calls are made through your existing Internet connection, there will be no more long distance call fees.  calls equally with a monthly fee.

You will find that prices for VoIP solutions vary widely since there are so many different ways to deploy VoIP. Some general things for you to consider are: 

  • Do you want to pay for everything up front? 
  • Do you want a monthly recurring charge, or do you want to pay for some of the system up-front and pay the rest off monthly? 
  • Make sure you know what your total cost of ownership (TCO) is before signing on the dotted line.
  • Download our buyer’s guide to learn more!

Get a free consultation and multiple quotes on service through VoIP Supply today! We will help your business find the ideal solution.


Come back for more VoIP questions and answers next time! If you have VoIP questions to ask us, please contact our VoIP experts today at [email protected] or at (866) 582-8591 

More from: Q & A Ying-Hui Chen

How to Configure Fanvil’s X2P Phones Manually

September 8, 2020 by Ying-Hui Chen

Fanvil’s X2P IP Phone is one of the most popular IP phones for call center environments. The X2P comes with an innovative design, high performance, and product efficiency.

Before your X2P can provide telephony service, at least one line must be configured on it.  The line configuration stores the service provider and the account information used for registration and authentication. When your X2P is configured, it will be able to register the device to your service provider with the right address and authentication. Let’s take a quick look at how this can be done!

How to Manually Configure a Line on Fanvil X2P IP Phone

To configure a line on your Fanvil X2P IP Phone, simply follow these steps:

  1. Open the line configuration screen through the soft-menu button: Menu > Settings > Advanced Settings 
  2. Enter the correct PIN code to enter advanced settings to edit line configuration. (The default PIN is 123) 
  3. Enter Accounts > SIP1 / SIP2 > Basic Settings
  4. See parameters and screens below, see all of them in the user manual here.

There you have it! Are you ready to learn more about the X2P? Visit our product page or call one of our VoIP experts today at 1-800-398-8647 to learn more!

How Many Simultaneous Calls Can a SIP Trunk Handle?

September 4, 2020 by Ying-Hui Chen

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. 

Q: How many simultaneous calls can a SIP Trunk handle?

A: You need one SIP Trunk channel for each simultaneous call. If your business typically requires five active calls at any given time, then five SIP Trunk channels are required.

VoIP Supply works with many popular VoIP service providers. We will work with you to determine how many simultaneous calls your business needs and make sure they are available to you. Call one of our VoIP experts at 1-800-398-8647 for a free consultation today!


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 

What Are the Replacement Options for The Cisco SPA112 VoIP Adapter?

September 3, 2020 by Ying-Hui Chen

As Cisco’s SPA112 2-Port Phone Adapter goes End of Life (EoL), many customers are asking for a replacement option. If you are also looking for a solution, here are a few options for you!

What replacement options are available for the Cisco SPA112 VoIP adapter?

1. Grandstream HT802 VoIP Adapter:

The HT802 is a Dual FXS Analog Telephone Adapter (ATA) designed for any residential or business setting that wants to turn their analog phone into a VoIP system. Our customers love how compact it is and the support of T.38 Fax over IP. See more features:

  • Two SIP profiles
  • Three way conferencing
  • Unique security certificate per device 
  • Auto provisioning options
2. Poly Obihai OBi302 VoIP Adapter: 

The Obi302 is a 2-port Analog Telephone Adapter (ATA) that allows you to use your existing analog phones or fax machines to make calls using the internet. The OBi302 includes a built-in 2-port router with QoS and comes with the ability to work wirelessly using OBiWiFi or OBiBT.

  •  2 Port FXS for one analog device
  •  Support for 4 SIP accounts
  •  Will support faxing using T.38
  •  Includes 2 RJ45 ports with integrated QoS 
3. Fanvil G200S VoIP Adapter: 

The G200S is a VoIP gateway and adapter that supports two SIP accounts and T.38 Fax over IP service. It is an affordable and convenient choice for connecting your existing analog devices to an IP phone system.

4. Patton SN102 SmartNode Analog Telephone or Fax Adapter

The SN102 is another great ATA alternative to the SPA112 that allows you to make VoIP calls with legacy devices. The SN102 is secure and reliable and it comes with advanced telephony features such as 3-way conferencing, call waiting, Do not disturb (DND) support an more.

  • Call Waiting, Call Transfer
  • Call Forward as Busy forward; Non-Answer forward; unconditional forward
  • Do-not-disturb (DND) support
  • 3-way conferencing

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