November 16th, 2018

11/16/20180 Comments

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Introducing the Panasonic Mobile Softphone

Panasonic has taken advantage of the incredible flexibility that IP telephony provides by introducing its new Mobile Softphone. It integrates with Panasonic IP telephony systems to provide versatile and adaptable unified communication services, anytime, anywhere.

Linking business with versatile mobility

Panasonic’s Mobile Softphone is an app that runs on any modern Android or iOS smartphone and allows users to carry their internal corporate extension number with them wherever they go. As long as the smartphone has data connectivity, that internal extension number exists on the phone, no matter where in the world it is located.

Employees can use the app just like they would use their smartphone to speak on the telephone. They can accept incoming calls to their internal corporate desk phone number, directly from their smartphone. They can also place calls from the app to any number and will be charged for the call as if they were physically in the office. All communication takes place over the smartphone data connection and not the telephony connection to the mobile operator, so no regular or roaming charges from the mobile operator are incurred.

Panasonic IP phone, IP PBX and Mobile Soft Phone

The softphone app functions as any other endpoint of Panasonic’s VoIP solution; no additional software or server is necessary. The app includes video calling capabilities as well as push notifications for incoming and missed calls, without the need for the app to be running all the time. This saves phone battery life, as well as CPU and memory resources.

Calls placed over insecure networks such as public Wi-Fi infrastructure or connections over the public internet are protected by using the industry standard of Session Initiation Protocol Transport Layer Security (SIP-TLS).

System requirements and key features

Panasonic’s Mobile Softphone for both Android and iOS is compatible with the KX-NSX2000, KX-NS1000, and KX-NS700 IP PBXs, and can be integrated directly into existing implementations with the appropriate PBX version and activation key. The app supports all of the most common – and some not so common – features, codecs and functionalities necessary for virtually all user categories.

Key features

Supports the SIP protocol, Call functions

* Make call

*Reject call

* Cancel call

* Video call

* Call hold

* Call transfer

* Blind call transfer

Supported audio codecs:

722, 729a, 711a/μ

Supported video codecs:

*264 Base Line Profile Encode up to VGA

*Decode up to 720p

*Security – industry standard SIP-TLS

*Push notifications

*Call log/missed call indicator

*Supports 21 display languages

The app can be downloaded directly from Google Play or the AppStore and can run on any device running Android 4.0 or later, or iOS 10 or later. These installations can then be activated by purchasing an activation key, which is offered as a single key or in bundles of 5, 10, 25, or 50 users.

Panasonic’s new Mobile Softphone is a secure, easy-to-operate smartphone app offering employees the freedom of mobility for voice and video calls, all from the convenience of their smartphones. In this way, it minimizes costs while providing mobility, security, efficiency, and flexibility to its users and their businesses.

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The 3 most common hybrid deployments for

8/3/20180 Comments

VoIP-to-analog solutions

Voice over IP delivers major improvements to traditional communication networks, and as a result legacy analog phone systems are on their way out. Ten years ago, 80% of homes had landlines, but today that number is down to 43%— and businesses are following this trend, as well. However much VoIP is becoming the go-to technology for communications, it is important to acknowledge that certain markets cannot fully convert to a total VoIP solution. Not recognizing this could lead to loss in potential sales against competitors. In these situations, a hybrid VoIP network can help deliver the flexibility and power of an IP solution while catering to the restraints of your customer. Below are some of the most common situations and their solutions.

Existing infrastructure investment

One of the most common reasons businesses may want to forgo VoIP is because of the existing investment in their telephony network. A completely new communications deployment may be something that can scare a company, especially small to medium-sized businesses who have limited resources. An easy way to work around this is to construct a hybrid VoIP network that maintains the initial analog investment while taking on the benefits that VoIP and SIP offer. An analog-to-VoIP gateway can connect to the legacy telephony endpoints while also connecting to an IP PBX to seamlessly integrate incoming VoIP trunks and an IP infrastructure with the existing analog devices.

Lack of VoIP availability

There are various reasons why businesses could lack the capabilities to switch to VoIP entirely. Geographically related bandwidth restraints or legal reasons could prevent businesses and enterprises from being able to utilize SIP trunks, restricting them to PSTN lines. In this case, a VoIP environment can be deployed for an internal network that connects to legacy landlines. An FXO gateway or ATA can convert PSTN legacy lines into the modern and more efficient VoIP network. This allows larger businesses to benefit from the productivity that VoIP brings to internal communications, while still being connected to the PSTN cloud.fxo-gateway-deployment

Mass PSTN to SIP Deployments

The final and most common reason for hybrid deployments is something many ITSPs worldwide have to address. Within the residential sector, many potential customers simply do not want to go through the hassle of transitioning over to VoIP because of the assumed difficulties of learning a new technology. Creating a simple transition for customers who don’t want to learn new technology and allowing them to continue using their analog device is the best way to go about tackling this obstacle. A simple one-line ATA is the best way to allow a customer to continue using their device without seeing any changes in how they actually make a phone call. On top of this, they get the savings and benefits that using a SIP trunk over a PSTN line delivers. ATA connecting an analog phone to the internet

CONCLUSION

VoIP is rapidly on its way to being the standard in communications, but there will always be situations where potential deployments could be delayed or declined due to PSTN-related reasons. Being able to clearly identify these roadblocks and understand how to construct a hybrid solution around them can be what makes or breaks a potential sale.

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When it comes to video conferencing, one size doesn’t fit all

3/14/20180 Comments

The GVC3200 series of video conferencing (VC) systems offers options that cover the diverse needs of businesses big, small and in-between. The series consists of three models: the GVC3200, GVC3202 and GVC3210. In this article we’ll explore each device in the GVC family and help you understand the flexibility and customization provided by each one — guiding you to choose a device that works the way your business does.

Meet the family – the GVC3200 series

The GVC series combines SIP, AndroidTM, H.323 and cloud conferencing capabilities into one device. What exactly does all this mean? Whether you want multi-point or point-to-point conferencing, video calls through your choice of popular Android apps like Skype, communication with third-party systems such as Polycom or Cisco, or high-capacity cloud conferencing, you can make it all happen with the GVC series.

When it comes to diverse video conferencing needs, the GVC series has you covered.

Need to conference with a client or office that has a different VC system?

Use either SIP or the H.323 protocol and done.

Want to host webinars for a large audience?

Use your GVC with IPVideoTalk or another cloud meeting platform.

Have attendees that can’t join from a VC system?

Bridge their Skype call into your GVC call and keep the meeting going.

Have a remote workforce?

Connect your conference room with users through their desktops, laptops and smartphones through the use of cloud meeting platforms. Our recommendation is Grandstream’s own IPVideoTalk.

On a tight budget?

The GVC series is designed to be accessible for any business.

You can see what we mean when we say the GVC series is flexible. Let’s meet each member of the GVC3200 series family to help you determine which one has the feature set for your business’s video conferencing style.

GVC3200 – The original VC system from Grandstream, the GVC3200 offers up to 9-way video conferencing through its built-in MCU (multi-point control unit), a powerful 12x zoom wide-angle camera, 3 monitor outputs and easy content sharing. It features up to 1080p video resolution and HD audio, making sure every conference maintains top visual and audio quality.

The GVC3200 is a VC system that is both powerful and flexible, with the capacity to keep up with the needs of a growing business and the flexibility to communicate on your terms. This system works well in larger spaces and conference rooms, and for users who prefer more screen layouts, it features 3 HDMI-out ports. The GVC3200 also has integrated Wi-Fi, allowing it to connect to your network without the need to install new Ethernet cables.

GVC3202 – The little sibling to the GVC3200, the GVC3202 offers the same flexible conferencing platform but in a lower-capacity package. The GVC3202 also has a built-in MCU and offers up to 3-way video calls with a 9x zoom camera, 2 HDMI-out ports for monitors, and omits the integrated Wi-Fi feature available in the GVC3200. This model is well suited for smaller conference rooms, personal offices and the increasingly popular “huddle-room” space. Users who plan to mainly use cloud conferencing platforms also prefer this model, as the cloud can host larger conferences beyond the GVC3202’s MCU, thus saving money by not purchasing a system with more native capacity than needed.

GVC3210 – Our newest addition to the family, the GVC3210 is a video conferencing endpoint made for users who can forgo an MCU and prefer to use point-to-point or cloud conferencing instead. At a market-leading price point of $999 (MSRP), the GVC3210 gives businesses the option to eliminate costly hardware and spend less on cloud services. As with the other models within the series, the GVC3210 can be used with Android apps like Skype and Google Hangouts, and supports the H.323 protocol, allowing users to call third-party conference devices.

Features of the GVC3210 include an Android 6.x operating system, built-in speaker, 4-MIC array and support for external audio equipment, a 90-degree wide-angle lens and 2 HDMI-out ports. The device can host point-to-point calls with 4k video resolution for the highest quality video experience. The GVC3210 is easy to install and a great endpoint for portable spaces, offering both Wi-Fi and Miracast, wireless screen sharing, as well as a magnetic monitor mount for installation without drilling or running new wires.

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hOW sip ENABLES WORKFORCE MOBILITY

2/14/20180 Comments

A decade ago, it would have been rare to meet someone who could carry out their profession from anywhere. For most, “going to work” involved physically going to a place of business. Today, because of the swift development of collaboration tools and the ubiquitousness of connectivity to the internet, working from anywhere is not only an option, but is often preferred by both employer and employee. Studies have shown that a mobile workforce tends to be more productive and experience higher job satisfaction.

Voice over IP (VoIP) technology is one of the major facilitators of employee mobility. Here we highlight some of the functionalities and tools that allow for a seamlessly mobile workforce.

SIP client on your phone. Both Android and iOS mobile phones have many free SIP clients available for download from their respective app stores. A SIP client allows you to register your mobile device on your SIP server (that is, your IP PBX) via your data connection or Wi-Fi and have the internal extension of your office desk phone ring on your mobile phone. You can also provide your mobile device with a unique internal extension, essentially supplying your office with an additional telephony device. With the appropriate network configuration and security, this extension can be used from anywhere in the world as long as your phone is connected to the internet.

Follow me. This is a feature that allows someone to call a single phone number and have that number ring either on your office desk phone, your mobile phone or any other phone you choose, depending on where you are at the time. This could function either as a hunt list, where each of these phones would ring in sequence if not answered after a predefined amount of time, or you could manually set your location so that the call will terminate on the appropriate phone. Some advanced systems can even use information from the GPS of your mobile phone to determine which phone should ring.

Teleconferencing. SIP is a protocol that supports both voice and video. As such, it can be integrated into a teleconferencing or even a telepresence system. With the appropriate hardware, a teleconferencing system can interconnect multiple endpoints regardless of the capabilities of each. For example, a videoconference can take place with one location having a full audio/video system, another participant using her mobile phone via the internet with both audio and video, a SIP phone user connected via voice only, and a PSTN user that has called in using his telephone.

Roaming headsets. Headsets are used with many types of devices, including mobile phones, desktop phones, computers, tablets, car audio systems and televisions, to name a few. A single wireless headset that can connect to all these devices can vastly simplify their use while providing a simpler teleworking experience.

Unified Messaging. This feature provides a single interface for accessing all types of electronic messaging including email, fax, text, voicemail and others. All of these messages can be sent via email, thus giving you only one interface to check for all of the messaging services. Adding features such as text-to-speech will allow you to literally hear your faxes or listen to your emails while driving, for example.

Hot desking. Closely related to extension mobility, this feature allows you to log into the IP phone at any desk and obtain your personal extension on that phone. This includes all of your settings such as speed dials, ringtones, number of rings before going to voicemail, and even the brightness of the display.

Security Considerations

It is important to note that, as useful as these functionalities are, they must be employed with caution. Because many of these features function outside of the relative safety of the corporate network, it is vital that the appropriate security measures be taken to mitigate potential security threats. These may include:

The use of VPNs to connect mobile phones that register with the SIP server through the internet.

The employment of the appropriate firewall configurations on the edge of the corporate network that will perform deep inspection of packets to verify their source and type.

The implementation of measures to avoid toll fraud from users of internal extension numbers on their mobile phones.

The configuration of encryption for all voice communications to eliminate the possibility of eavesdropping, either from inside or outside of the corporate network.

CONCLUSION

VoIP technologies enable a more productive, more mobile workforce in multiple ways. As more and more employers realize the benefits of working remotely, we can expect a steady increase in the implementation of these technologies worldwide.

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Common VoIP issues and how to fix them

2/9/20180 Comments

For networking professionals, one of the most difficult things to deal with is troubleshooting VoIP issues, because the troubleshooting process for VoIP is not always intuitive. Answers to the questions, “What could be wrong?” and “What should I check first?” are not always readily apparent.

To aid your troubleshooting and allow you to take some meaningful actions before reaching out to your telco or vendor help desk, we’ve listed the most common VoIP problems you may face, as well as their causes and solutions.

Voice and data are different

Although it uses the same infrastructure as conventional data communications, voice over IP behaves differently on the network and has very different requirements. This is why it is often challenging for those accustomed to troubleshooting data network faults to pinpoint problems that may arise with voice services, even though the network seems to be working “just fine.”

The good news is that today, VoIP is a mature and dependable technology whose kinks have largely been ironed out and dealt with. Chances are that any problems you may face have already been faced and solved by other networking professionals in the past.

Blocked ports

Some of the most common VoIP issues involve the blocking of TCP and UDP ports. Ports are the addresses employed on the Transport Layer of the OSI model that are used on a device to distinguish between applications and services. Various voice services use specific ports to function. If these ports are blocked at any point between the communicating devices, voice services may fail partially or completely. Depending on what ports are blocked, different functionalities of VoIP will be affected.

The location on the network where ports are most commonly blocked is at the network edge; that is, the point where the enterprise network meets the ISP (internet service provider) and the internet. At this location, there may be several mechanisms being employed such as access lists, firewall rules, or network address translation (NAT) that may be responsible for blocking ports. These services are all vital to the functionality and security of a network; however, they can also be the cause of VoIP failure.

Access lists (ACLs) – ACLs are rules found on the edge device of an enterprise network (a router or a firewall) that block or allow packets based on their source and destination IP addresses and ports. If ports that VoIP services require are blocked, then calls or registration may fail.

Firewall rules – Firewall rules go one or more steps beyond simple ACLs. Firewalls are able to inspect each packet that attempts to enter the enterprise network and to decide, based on specific security policies, which packets will be allowed, and which won’t. Other than source and destination IP addresses and ports, firewalls can look deeper into a packet and determine if it is safe to let the packet through or not. If a firewall is not configured to allow voice services to pass, a failure can occur.

Network address translation (NAT) – NAT has been the great deliverer when it comes to delaying the inevitable exhaustion of IPv4 addresses. By providing a translation from internal private IP addresses to external public IP addresses, it allows for the reuse of IP addresses within enterprise networks without any danger of conflict, thus giving several more years of life to the IPv4 addressing scheme. At the same time, it can be a nightmare for VoIP as it often causes problems with voice calls, especially those that are initiated from the outside. Elaborate best practices have been devised and have even been written up in an RFC to define NAT traversal practices for SIP-based voice communications.

Voice services affected by blocked ports

Different ports being blocked may impact voice services in different ways. This depends on what part of a voice service is being blocked.

Signaling - VoIP most commonly uses the Session Initiation Protocol (SIP) for signaling. What is often misunderstood is the fact that SIP does not carry any voice packets. It is used to initiate a call, send ringtone, provision bandwidth, and to establish services like conferencing, call waiting and call park, to name a few. If SIP ports are blocked, no calls can be initiated, the IP PBX cannot register with the SIP trunk, and telephony endpoints cannot register with the IP PBX. The default ports that SIP uses are 5060 and 5061.

Voice traffic – IP telephony traffic is carried by Real-time Transfer Protocol (RTP) and is monitored by RTP Control Protocol (RTCP). RTP and RTCP typically use UDP ports somewhere between 1024 and 65535. The specific ports that are used depend on the configuration of each IP PBX, IP phone and SIP trunk, and are largely defined by each individual vendor and telco in their own systems. If these ports are blocked, a call may actually connect successfully using SIP, but may experience one-way or no-way audio.

Trunk provider ports – Ports must be forwarded correctly both on the network edge and on the SIP trunk provider’s end of the link. Whenever a VoIP dysfunctionality is detected, the correct forwarding of ports should also be checked.

Quality of Service

Another set of voice-related malfunctions are linked to the amount of traffic on the network. A converged network is one where both voice and data traffic share the same infrastructure. Initially, voice services may function with an acceptable level of quality, but as data patterns and data applications change on the network, degradation can occur. This can result in delays in voice transmission, intermittent interruptions, the introduction of jitter and a general decrease in call quality. This is most likely due to either the lack of Quality of Service (QoS) mechanisms on the network, or their inadequate or faulty configuration.

Firmware

Finally, many issues with VoIP are related to how the operating systems and firmware of VoIP devices and servers operate. Vendors continually update their firmware and provide free access to the most up-to-date versions online. IT professionals should upgrade and update device firmware regularly to make sure that any bugs or faults discovered by other users can be proactively dealt with and resolved.

If you still have to call tech support…

Even if you do proceed with troubleshooting, there are times when reaching out to the vendor or the telco is necessary. Keep in mind that if you have done the proper preliminary steps and you have gathered the appropriate information, you can substantially cut down on the time it will take the trained techs to solve your issues. When making the call, be sure to have the following information ready:

If a user is complaining about faulty behavior of the telephony system, try to reproduce the fault so you have firsthand experience with it, verify that it is not user error, and be able to better articulate the issue to tech support.

Have the firmware and/or software version numbers of the IP PBX, gateways and any other network equipment handy.

Have the telephone or softphone model numbers or versions along with their configuration and setup on hand.

Depending on the policies of the help desk you are contacting, be ready to provide some supervised access to your system via a remote desktop application.

CONCLUSION

Even though IP telephony shares the same infrastructure as conventional IP data networks, its behavior and troubleshooting methodology is far from similar. For this reason, it is important to be able to identify the problems related to VoIP and to determine more precisely where the fault exists, so it can be pinpointed and resolved more efficiently.

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How to hack-proof your VoIP network

1/31/20180 Comments

One of the most common matters we discuss with customers on our support calls is network security – specifically, how to lock down SIP Trunks, IP PBXs, SIP phones, and routers from hacking. Here we review different ways hackers can break into your voice network and the steps you can take to secure your system.

VoIP vs. traditional telephony

Traditional voice systems were notoriously difficult to hack into. Attackers would have to gain access to proprietary systems from very few available entrance points to compromise them. Physical presence would usually be necessary either from within the telco’s central office or from within the cable closets of an enterprise to eavesdrop on voice conversations or to commit toll fraud or other harmful actions. A strong physical security in both of these areas was almost always enough to protect a voice network from all but the craftiest of attackers.

Enter Voice over IP. VoIP technology takes advantage of the infrastructure of IP packet networks. This has allowed for advanced voice and unified communications (UC) services that were impossible with traditional telephony. At the same time, it opens up voice networks to the same inherent security risks that IP packet networks have.

The good news is, these security risks can be mitigated. To do so, it is important to identify and understand the risks as well as the methods used to minimize them. With these security precautions, the advantages of VoIP handily outweigh the extra effort of securing them from potential attackers.

Voice over IP threats

The following are some of the most common threats to VoIP systems:

Identity and service theft – Attackers may attempt to steal services from a service provider while charging the cost to a third party. For example, an attacker may obtain access to a third party’s SIP trunk and use the subscriber’s credentials to initiate their own calls. This not only incurs costs to the legitimate SIP trunk subscriber, but also allows the attacker to take action on the SIP trunk using the subscriber’s identity.

Eavesdropping – Another common threat to VoIP systems is eavesdropping. With traditional telephony, it was necessary to gain physical access to the wires of a specific telephony circuit to perform wiretapping to eavesdrop. With IP telephony, packet sniffer software can be used to capture voice packets and reassemble them to hear the conversation that took place. Conversations need not be listened to in real time but can be saved and listened to at a later date. This type of attack is similar to the Man-In-The-Middle (MITM) attack that can occur on IP packet networks.

Viruses and malware – Because VoIP exists within the realm of computer networks, it is vulnerable to the same types of attacks as computers are – from viruses and malware. VoIP systems such as IP PBXs, IP Voice servers and even some kinds of IP telephones may be vulnerable to viruses that are specially designed to infect voice systems. Malware can also install itself onto these systems and compromise their functionality.

Call tampering – This type of attack involves the tampering of a phone call in progress. This can result in degraded quality of a call, interfering with SIP signaling, or the complete disconnecting of a call. Its purpose is to deliberately cause low-quality voice network performance.

Threat mitigation

These are best practices for minimizing the vast majority of VoIP network attacks:

Encryption of voice packets and signaling – The application of an encryption algorithm on all voice packets and SIP signaling packets is a best practice that should be observed by all VoIP professionals. Almost all commercially available VoIP equipment, including IP PBXs, gateways, IP phones and softphones, provide some level of encryption that is relatively easy to implement. The most powerful encryption algorithms available today include the Advanced Encryption Standard using key sizes of 256 bits. Such encryption will mitigate against eavesdropping as well as identity and service theft.

Encryption is a service that SIP trunk providers often apply to voice packets, but less often on the SIP signaling packets. It is a good idea to make sure your SIP provider implements both and that your voice gateway and/or IP PBX supports the encryption methods used by the telco.

Conventional IP network security – For the most part, measures that are considered best practices for mitigating the risks inherent to IP networks are sufficient for protecting a VoIP network. Antivirus and anti-malware software, firewall protection, DDoS attack protection, as well as VLAN isolation and subnet segregation, all contribute to the security of VoIP networks. This is because the VoIP and IP data networks are one and the same. These security measures will reduce call tampering as well as unauthorized access into VoIP systems and devices.

Our article on how to keep your SIP phone system safe has more details on how to mitigate specific types of risks, including DDoS attacks, packet sniffing, data extrusion and malware.

Physical security – Physical security is just as important for modern VoIP networks as it was for traditional telephony networks. Although remote attacks are more common on VoIP networks, lax physical restrictions to telecom closets can still result in VoIP network compromise.

For more details on how to beef up the physical security of your network, see Five steps for securing your SIP telephone system, as well as our article on securing your wireless network.

Securing specific elements of a voice network

There are specific steps that you can take to secure different parts of your voice network.

IP Phones – The IP phones themselves should be protected from physical access to avoid unauthorized calls being made, especially during off hours. Additional security can be added using authorization codes that must be dialed before each toll or international call or even a general security lockout code that will make the phone inoperable until a PIN has been entered. Additional network-level security that can be implemented includes encryption of voice packets to and from the IP phone, segregation of voice on a separate voice VLAN, and the regular updating of phone firmware to resolve any security holes that may have been discovered.

IP PBXs – Most IP PBXs run on either Linux-based or, to a lesser extent, Windows-based servers. The usual best practices employed for these servers, such as antivirus, software firewalls, malware protection and regular backups of vital data, are ordinarily sufficient to provide the required security for IP PBXs. Other IP PBXs may run on appliances that resemble a router rather than a server. These devices must be approached more like network devices rather than Windows or Linux servers. See the next section on routers for this.

Routers – Network devices such as routers should be secured using the typical best practices used to harden such devices. This includes password protection, encrypted management communication, access lists which block specific IP address ranges and Transport Layer ports, as well as shutting down any services that are not in use. These best practices also apply to any router-like VoIP appliances such as IP PBXs or voice gateways.

SIP Trunks – When a voice conversation takes place over a SIP trunk (or between any SIP-enabled devices, for that matter) the voice packets and the SIP control packets that are exchanged form two separate and independent sessions or channels between the participating SIP endpoints. Encryption is a service that SIP trunk providers often apply to the voice portion of the communication, but not as often to the SIP signaling and control packets. It is a good idea to make sure that your SIP provider implements encryption on the SIP messages and that your voice gateway and/or IP PBX supports the encryption methods used by the telco.

CONCLUSION

Securing a VoIP network is often a simple matter of implementing some basic network security best practices. Because of network convergence, most of the vulnerabilities to the VoIP network can be mitigated by the same practices used to secure IP networks. The extra effort involved in locking down a voice network is vastly outweighed by the immense benefits it offers in the form of more advanced and cost-effective voice services, UC services and enterprise productivity.

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Unified Communications for small to medium sized businesses

1/2/20150 Comments

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Unified Communications (UC) is one of the fastest growing technology platforms active today. And its no surprise why , with the growing trend of remote works, ever increasing mobile demands, and the ease of use of small screen and tablet applications. All of these factors add up to the same conclusion - companies are realizing they need a phone system platform that can meet the demands of today's high-tech needs.

Even if you are not totally sure what Unified Communications (UC) is or haven't heard the term before, you are probably somewhat familiar with the benefits. In fact, if you have ever received a txt msg of a voice mail, or had a phone call forwarded - then you have received some of the benefits this amazing technology has to offer.

Whether your company employees remote workers, staff that spends a lot of time in the field, or simply personnel that is constantly in and out of the office, your company can greatly benefit from a Unified Commutations platform by Panasonic.

So what exactly is Unified Communications any way?

Well, simply put, Unified Communications is the integration of real-time communications instant messaging with non-real-time services like email or voice mail. The magic of how the UC technology works lies in the amazingly broad abilities of internet based phone systems like the KX-TDE200 by Panasonic.

Benefits of Unified Communications for the company:

Time is money - UC enables employees and key staff to receive and respond to voice mails, text messages, and emails in real time.

UC allows you to integrate voice and data communications, freeing up resources and consolidating your needed equipment

UC increasing productivity of key personnel and reduces complexity and technology hurdles

In a nutshell, anytime you can streamline your technology and make employees more productive, you are going to increase bottom-line profits.

If you are interested in learning more about how a Unified Communications solution by Panasonic can help your company, send us an email or fill out the contact form on our website.

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0% Financing For Panasonic Business Telephone Systems

8/27/20140 Comments

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Another Great Reason To Choose Panasonic Business Phone Systems

Action Phone Systems is Houston's #1 Panasonic Reseller. Now through 09/30/2014, we are offering an EXCLUSIVE 0% financing option for Panasonic Business Telephone Systems.

A good business owner knows that having state-of-the art equipment can make a day flow smoother, more productively and can increase revenue. But sometimes the upfront cost of investing in this necessary expense can really hurt!

What are the advantages of leasing?

Here are a few great reasons why leasing your Panasonic Phone System makes good business sense:

You can preserve your working capital

You can have fixed monthly payments

100% Financing

Promotion Details

0% Interest, 36 month finance term

Zero down/purchase option = fair market value

Offer valid on entire Panasonic Business Telephone System portfolio

Equipment can be purchased at the end of the lease term for a fixed market value rate of 20% of the original equipment cost

Other terms are available upon request

This great offer is only good through 09/30/2014. Contact us today to get more information and order your new equipment here.

(Each transaction must be credit approved. Rates subject to change. Valid on finance agreement closed between 07/01/2014 and 09/30/2014. Certain equipment restrictions may apply. Not applicable for organizations that have been in business less than 2 years. This promotion can not be utilized in conjunction with any other promotion being offered by Panasonic Business Finance.)

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Moving A Business Phone System In Houston

7/1/20140 Comments

If you're one of the 1,000s of companies this year that is moving your office, then you already know the near inexhaustible to-do list that is ahead of you. While we can’t help you with the other items, we have put together this easy-to-follow checklist to make moving or switching your phone system in Houston easier.

Phone System Moving Checklist

Contact your telephone provider & internet service provider (ISP) and verify availability of services. Plan out your installation in advance to account for potentially long lead times; some areas & some service providers can take several weeks.

After you have made arrangements with both your ISP and/or your phone service provider, contact your company’s business phone system provider in Houston to make necessary arrangements to have your phone system moved. More than likely, at this time, you will need to provide a few key pieces of information to them like:

• Location of the new office
• Square footage of the new office
• Number of extensions needed
• Number of cable drops necessary
• Will you have a dedicated are for the phone system & auxiliary equipment
• Etc…

If you have the architectural drawing of the office space, then it is a good idea to go ahead and mark off where the cable drops, phone lines, and voice/data equipment will be. Giving your phone system provider in Houston a copy of this drawing will be very helpful.

Finally, you will want to plan for the “gap of service” that applies to when you vacate your old office and you are set up in your new office and the phones go live. There are a number of ways to handle this and each situation may require a different solution.

For instance, if you are switching numbers completely (relevant if you need to leave the area code), then you can contact your telephone service provider about setting up a reference message for the first 6-12 months after you change numbers. In other situations, you can simply have the number forwarded or if your company utilizes a VoIP phone system, and then it can be as simple as getting the data connected at the new offices and transferring the number/s on a certain date.

Important Note: Make sure that the data is up and running and everything is in place prior to transferring for all VoIP!

Regardless, what type of phone service your business uses, or who your phone system provider is, you can take the worries of moving your phone system in Houston off the list by using our checklist and keeping your phone system provider in the loop.