Although small businesses can get numerous benefits from using an appropriate VoIP Phone System. Here we have listed the 7 top Small Business VoIP Benefits you should know as a small business owner.

1 – Save Money

Initial investment, installation, ongoing management and maintenance all contribute to a much lower total cost of ownership for VoIP Phone Systems as compared to Legacy Phone Systems for even a basic feature set. Cost Savings begin with the consolidation of data and voice onto a single network. The savings continue with simplified management and maintenance of the network and phone system. VoIP Phone Systems also enable savings in other areas including lower phone bills and more productivity.

2 – Enhance Productivity

VoIP opens up an array of new productivity enhancing features to make communication more efficient. For example, features like Unified Communication, Virtual Numbers, Voicemail to Email, User Presence, Find-Me Follow-Me, Web integration, etc. are only possible on a VoIP Phone System.

With unified messaging, voicemail messages are delivered to user’s email inbox. 3CX Phone System for Windows is a revolutionary IP PBX that delivers unified communications technology by merging voice and data networks, allowing businesses to simplify real time delivery of information and ensure ease of use.

3 – Easier Management

Many VoIP systems offer a user-friendly web-based interface for managing users, call settings, and features. This greatly reduces the need for expensive support calls and site visits. Many systems are even remotely configurable and manageable. So you can do it all yourself with relative ease or let a VoIP professional do it all remotely. That’s right, no truck roll!

4 – Business System Integration

Many VoIP systems can be integrated with business applications like Customer Relationship Management and Contact Management software such as Salesforce.com, Microsoft Dynamics and Microsoft Outlook to name a few. This enables features like Click to Call and Screen Pops. For example, a Sales Rep picks up a call and within seconds the caller’s order history pops up on the screen enabling the Sales Rep to see the customer’s past buying history and make the right pitch. This greatly helps enhance the customer’s experience resulting in higher satisfaction and more repeat sales.

5 – Grow Easily

Since many VoIP Phone Systems are now software based, they are very easy to grow even if system capacity is exhausted. Simple additions like adding phone lines, extensions, or features on a VoIP Phone System is very easy. In most cases, this can be done with a few mouse clicks. This ensures that no matter how your business grows, you can quickly scale your VoIP Phone System to avoid lost opportunities.

6 – Built-in Mobility

Most VoIP Phone Systems offer some form of mobility. Users can take the office wherever they go by accessing their VoIP phone from any Internet connection. Mobile users can check voicemail via email, call company extensions for free, and make long-distance calls from the work extension instead of the mobile. For example, the 3CX Windows Phone System offers 3CX Phone, a phone app that can run on iPhone, Android and Windows to allow you to take your extension with you virtually anywhere.

Another aspect of mobility is when you have to move your office or room to a new location. Moving or re-arranging desktop phones within an office is simply a matter of unplugging the phone set and moving it to a different network outlet. When moving to a new location, it is also much easier to move a VoIP system as there is usually no need to involve the Telco, and pay for technician site visits and rewiring. Also phone number limitations due to location are a non-issue with VoIP.

7 – Telco Friendly

If you are concerned about existing phone lines and more importantly your phone numbers that are published everywhere, there is a solution. VoIP Phone Systems can be Telco friendly to allow you to continue to use existing land lines with your new VoIP Phone System. VoIP Phone Lines are also available that will allow you to cut the cord with the Telco while retaining your existing phones numbers. VoIP does not have to be an all or nothing solution.

If you are still not sure that a VoIP Phone System can really benefit your business, please do not hesitate to contact us for further assistance. We have an exclusive offer for small businesses who want to switch to VoIP Phone system (with enterprise features) for as low as $1785 or $70/month with one year long distance included.

The most efficient and economical way to add call center features to the customer support process is to integrate your existing VoIP/PSTN telephony system / PBX/ IP PBX/ PABX to your call center software.

3cx call center edition module

3CX provides one of the best VoIP Phone system software used by thousands of companies worldwide. Although most of the call center modules can work with any VoIP Phone system and VoIP IP PBX but the good news for all 3CX phone system users is that 3CX has launched a 3CX Call Center module a while back. This 3CX call center edition provides professional call center features at an affordable price, thus allowing you grow your business by focusing on providing the best possible customer service. The 3CX Call Center edition comes as a license key upgrade to 3CX Phone System. Simply reactivate your 3CX Phone System and bring to life call center features.

Key 3CX Call Center Features:

3CX call center edition comes with a complete range of important call center features required to give excellent customer support. However we are just listing some of the key 3cx call center features below.

Advanced Real time Queue Statistics

• Supervisors & Management get access to advanced real time status information in the Queues page of 3CX MyPhone.
• Number of callers in the queue, the number of answered/unanswered calls, the average and longest wait time, time an agent logged in/out of the queue and more viewable by supervisor.
• Calls waiting longer then a specific period of time can be taken directly.

Advanced Agent Statistics

• Log agents in and out of queues
• Time an agent logged in/out of the queue
• Review the number of answered/unanswered calls
• Average and longest wait time and more

Call Back Feature

• Allow your customers to hang up and retain their position in the queue
• The customer is called back when an agent becomes available
• No need for customer to wait on the phone to be served
• Call back notification emails are sent to the supervisor

Additional Queue Strategies

• Round Robin – Each call is distributed in a round robin fashion, i.e. first call will go to Agent 1, second call will go to Agent 2, third call will go to Agent 3, and so on.
• Longest Waiting – Next call will go to the agent, who takes the longest time to answer a call
• Least Talk Time – Next call will go the agent, who has the least total talk time. In this way, talk time is distributed evenly among the agents.
• Fewest Answered – Agent, which has answered the fewest calls is targeted in this strategy.
• Hunt By Threes – Random – 3 agents are selected randomly for the calls. If they don’t answer, then other 3 agents are targeted randomly.
• Hunt By Threes – Prioritized – Calls are sent to 3 predefined agents. If the first 3 do not answer, then the next 3 will be targeted.

SLA Alerts

• Supervisors and managers can be notified when callers have to wait beyond a configurable amount of time
• Events are logged in order to meet customer service requirements
• SLA notification emails are sent to the supervisor

Listen in

• Supervisors can screen calls by listening in without the agent and caller knowing about it

Listen and Whisper

• The supervisor may need to relate some important information to the agent without the customer hearing

Barge in

• The supervisor can enter the call and assist the agent and customer

3cx call center edition module features

The 3CX Call Center Module provides small and medium-sized companies with an opportunity to offer superior customer service without necessity to purchase an expensive enterprise call center.

To read a detailed review on 3CX Phone system for windows please visit our earlier blog post “3CX Phone System for Windows Review“.

3cx Certified Partner & 3CX certified VoIP service provider

To purchase 3CX Call Center Edition, visit our 3CX Phone System for Windows products page.

CiGear is a 3CX certified premium partner and 3CX certified VoIP service provider. We provide 3Cx Call Center Edition Products & 3Cx Call Center module Upgrade from small size of 4 simultaneous calls mini call center solution to 512 simultaneous calls enterprise edition.

For help with upgrading your existing 3CX license or deploying a complete 3CX call center solution, contact us today and we will discuss how we can help you communicate better, be more mobile, save money and grow your business!

Before I start explaining what is a Session Border Controller, let me first explain the architecture of a typical VoIP network, which will help you visualize and understand the purpose and functionality of SBC easily.

In a typical company, each employee has an extension on his/her desk connected to an IP PBX or phone system. The phone system may be connected to the outside world using a combination of analog or digital telco lines and VoIP. If a person dials a local extension, the IP PBX diverts the call internally to that extension. If the number dialed is on the public telephone network or PSTN, then that call is routed to the PSTN network. Since long-distance calls can be expensive, it is usually cost advantageous to route the call to a VoIP service provider instead of the local telco line to reduce the phone bill of the company significantly. Therefore, a device known as a Session Border Controller or SBC is deployed on the border of company’s private network and VoIP service provider’s public network.

What is a session border controller?

While sitting on the border of public and private network, the Session Border Controller or SBC can perform the following functions:

1. The SBC can protect the private network and other devices from malicious attacks such as a Denial-of-Service attack (DoS) or Distributed DoS, Toll Fraud via rogue media streams. It can also provide Malformed packet protection and perform encryption of signaling and media for privacy and security.
2. The SBC can allow different parts of the network to communicate through the use of a variety of techniques such as NAT traversal, SIP message and header manipulation, IPv4 to IPv6 interworking, VPN connectivity, Protocol translations between SIP, SIP-I, H.323.
3. The SBC can provide QoS benefits to a network including functions like traffic policing, resource allocation, call admission control, etc.
4. Session Border Controllers also often provide support for regulatory requirements such as emergency calls prioritization and lawful interception.
5. Some newer SBCs also have built-in digital signal processors (DSPs), through which they can have greater control over media/voice and can provide services such as DTMF Relay and Interworking, Media Transcoding, Tones and Announcements, Data and Fax Interworking, etc.
6. Finally, since all outside communication occurs through the SBC, it is a natural point to gather statistics and usage based information on these sessions.

The most robust and reliable Session Border Controllers are the Patton Smartnode Session Border Routers including the SN5200 and SN5400 product series. These products combine Session Border Controller and Router into one customer premise device.

Patton Smartnode 5200 Session Border Router:

PATTON SESSION BORDER ROUTER - 32CH - EUI

The SmartNode 5200 connects the Enterprise’s LAN to an Internet telephony service provider (ITSP), creating a single conduit for multimedia components including voice, video and data.

The SN5200 includes SIP-aware NAT and access controlled lists (ACLs) for maintaining secure communications when SIP traffic crosses the Enterprise edge.

SN5200 provides advanced features like NAT/ NAPT in addition to ACLs and PBR for security and QoS that keep malicious traffic from affecting bandwidth requirements of PBR-marked essential traffic. Downstream QoS ensures important voice and fax calls don’t get bogged down by such resource hungry lower priority TCP traffic as FTP downloads.

Patton Smartnode 5200 Session Border Router

Patton Smartnode 5400 Session Border Router with Transcoding:

SMARTNODE ENTERPRISE SESSION BORDER ROUTER - 2X GIGE - 32 SIP

The SmartNode 5400 series adds Media Transcoding features to the Session Border Router.
The SN5400 enables transcoding for up to 64 sessions. Most VoIP traffic within Enterprise LANs is uncompressed whereas traffic on the WAN is often compressed to save bandwidth. SN5400 transcoding between any VoIP codecs enables this optimal utilization of transport network at both sides of the Enterprise border.

Below is given a typical deployment of Patton SmartNode 5400 Session Border Router.

Patton Smartnode 5400 Session Border Router with Transcoding

VoIP Gateways

So if all this is done by a Session Border Controller or an SBC, you might be wondering what is the job of a VoIP Gateway. A VoIP Gateway is simply a translation device, which converts analog and digital telco lines to VoIP and vice versa. For example, if you have traditional telco phone lines and you want to use them with a VoIP phone system, you will need a VoIP Gateway.

The main functions of a VoIP Gateway include voice and fax compression or decompression, control signaling, call routing, and packetizing. High-end VoIP gateways also have additional features such as interfaces to external controllers like Gatekeepers or Softswitches, network management systems, and billing systems.

Some popular VoIP gateways are mentioned below:

Patton Smartnode 4110 Series Analog VoIP Gateway:

SMARTNODE 4 FXO VOIP GATEWAY - 1 X 10/100BASET - H.323 / SIP

The business-class SmartNode 4110 VoIP Media Gateway supports up to eight transparent phone calls and leverages VoIP for carrier and corporate access. Connecting to any analog phone, fax, or PBX, the SN4110 is an effective and flexible solution for toll-bypass, remote/branch office voice connectivity, and enhanced carrier services.

Patton’s SmartNode 4110 delivers the legacy phone interfaces, service transparency, and flexible PSTN integration for true converged packet voice.

Patton Smartnode 4110 Series Analog VoIP Gateway

AudioCodes MediaPack 118 Analog VoIP Gateway:

AUDIOCODES MEDIAPACK 118 ANALOG VOIP GATEWAY - 8 FXO PORTS - SIP

The MediaPacks are well-suited for commercial VoIP deployment.

Field-proven voice and fax technology and feature-rich design makes the MediaPacks an excellent solution for various emerging VoIP applications. The MediaPacks are used in VoIP-based PBX or IP-PBX architecture as a mediation gateway and remote analog extension.

It is also used for optimizing voice transmission over satellite links, payphone networks, centralized IVR and Quality Monitoring applications.

Mediatrix 3531 Single T1 Digital VoIP Gateway:

MEDIATRIX 3531 SINGLE T1 DIGITAL VOIP GATEWAY

Designed specifically for enterprise applications, the Mediatrix 3500 Series gateways make use of existing broadband access equipment to connect to any standards-based VoIP network.

The Mediatrix 3500 Series gateways meet the requirements of enterprises that want to connect their ISDN equipment, such as PBXs, through a T1 PRI interface to an IP network or as a gateway to the PSTN.

The Mediatrix 3500 Series offers security features such as TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signalling and media transmission aspects.

SmartNode 4940 T1/E1/PRI Enterprise VoIP Media Gateway

SMARTNODE HI-CAP 1 T1/E1/PRI VOIP GATEWAY - 1X GIGE - 15 VOIP CH

Providing a high-density seamless link between the circuit-switched telephone network and voice-over-IP, the SN4960 is ideal for PBX business trunking or corporate VoIP access. Offering up to four software configurable T1/E1/PRI interfaces the SN4960 connects to any switch, PBX and data network with up to 120 simultaneous calls using SIP, T1, E1 or PRI signaling. With its built-in CSU/DSU, any T1/E1 port can be selected as a WAN port for truly integrated voice and data access.

Like every SmartNode, the SN4940 delivers toll-quality voice with all industry standard CODECs including low-bandwidth G.723/G.729. Business class services are supported with T.38 fax, fax bypass and modem bypass features.

SmartNode 4940 T1-E1-PRI Enterprise VoIP Media Gateway

I hope you have enjoyed this article on Session Border Controller & VoIP Gateways. You can subscribe to our RSS feed to receive more informative articles and promotions from CiGear.com, your trusted partner for VoIP Phones, VoIP Phone Systems and all sort of VoIP Equipment.

There are many suitable VoIP Wireless phone solutions available depending on scale, coverage and environment involved. Before choosing a VoIP Wireless Phone, it is a good idea to define your requirements.

A typical VoIP Wireless Phone

- Who will be using the VoIP Wireless phone?
- Where will they be using it?
- What times of day and for how long will it be used?
- What features are required on the VoIP Wireless phone?
- Is there any wireless infrastructure already built that should be leveraged?
- What is the budget?

Since most of our readers are small and medium sized organizations, we will focus this article on solutions for small and medium business VoIP systems.

There are 4 options we will look at:

- VoIP DECT Phones
- VoIP WiFi Phones
- Analog Cordless Phones
- Analog Long Range Wireless Phones.

Snom M9 - VoIP DECT Phones

VoIP DECT Phones

Anchor textDECT is a digital technology that was specially designed for cordless phones so it gives you better battery life and in many cases better audio. VoIP DECT Phones typically have feature keys that can integrate with your VoIP Phone System to make it easy and convenient for users to hold, transfer, toggle and conference calls. There are a number of VoIP DECT products available. The products targeted for small and mid-sized deployments generally come with a base station that is good for unto 9 wireless phones and has an indoor range of about 150 feet. Repeaters are available to extend the range. Typical DECT handset battery standby time can be upto 100 hours.

In terms of cost, VoIP DECT phones work out to be less expensive in smaller areas. If you have a large area over which you need to extend DECT coverage for a small number of users, the cost can easily add up. A couple of popular VoIP DECT Phone models are the SNOM M9 and the Aastra MBU400.

cisco-wip310_4 - VoIP WiFi Phones

VoIP WiFi Phones

WiFi is a popular wireless data technology that is typically available at most businesses. VoIP WiFi Phones do not require a base station since they work with existing WiFi networks. VoIP WiFi handsets also usually have feature keys that can integrate with your VoIP Phone System to make it easy and convenient for users to hold, transfer, toggle and conference calls. Quality can depend on the quality of the Wi-Fi signal. Typical WiFi phone battery standby time is unto 40 hours. All of these features make this an ideal Business VoIP Phone to be used in the offices with ease.

VoIP Wi-Fi phones are more expensive than VoIP DECT phones but since there is no base or repeaters to purchase, overall cost is manageable especially if the WiFi network is already built and coverage is needed over a larger area. One of our most popular WiFi phones is the Cisco WIP-310.

cisco-PAP2TNA - Analog Cordless Phones

Analog Cordless Phones

With so many Analog Cordless Phone models on the market, this is likely the least expensive cordless solution for VoIP. To use an analog cordless phone with your VoIP Phone System, a VoIP Adapters is needed. The drawback of using an analog cordless phone with a VoIP adapter is that lack of feature buttons on the phone. So if all you need is to dial and receive calls, this may be a cost effective solution for personal or business usage.

There are many VoIP Adapters available to be connected with an Analog Cordless Phone. One of the most popular being the Cisco PAP2T gives best value for your money.

DURAFON-4X-PIA - Analog Long Range Wireless

Analog Long Range Wireless

Analog Long Range Wireless Phones are worth considering if your application is primarily in a factory, warehouse or campus environment where long range and durability are extremely important. This solution would also require a VoIP Adapter to integrate with your VoIP Phone System. However it is likely the best and possibly the only reliable long range solution for VoIP, albeit the most costly solution.

Our most popular long-range models are the Engenius Durafon and the Engenius FreeStyl. They can support from 1 to 4 incoming lines and up to 36 handsets.

For more information on product options, check out our VoIP Wireless Phones page or feel free to call us at 1-866-924-4292 for further help.

3CX is a software based communication solution that runs on Microsoft Windows. It is designed to replace the traditional hardware PBX or phone system. This award winning technology raises the bar for what you would expect from your phone system.

Technology behind 3CX Phone System for Windows

The technology used in 3CX is based on the popular SIP standard. This means it supports most popular VoIP phones, VoIP gateways, and VoIP services. This positions 3CX not only as an excellent VoIP system but also one that can communicate using traditional analog and digital landlines provided by your telco. VoIP Gateways enable 3CX to connect to your PSTN phone lines. 3CX offers built-in for VoIP lines from virtually any VoIP service providers globally without the need for any hardware. This makes it a truly versatile phone system solution.

Major Advantages of 3CX Phone System

One of the highlights about 3CX phone system that separates it from its competitors is easy management. System administrators will find 3CX to be quite intuitive and very easy to manage. The phone system can be run on virtually any Windows environment making it simple and easy to manage and maintain everything on a common platform.

Integration possibilities of 3CX Phone System

3CX can be integrated with Microsoft Exchange, Microsoft CRM, Salesforce.com and many other related business applications. It also offers a plethora of advanced features like Call Queues, Conference Bridges, Fax Server, Skype support, Hotel Module, Call Center Module, Paging and the Voice Application Developer kit (VAD). Each of these topics would merit a separate article if not a chapter in a book!

Pros of using 3CX Phone System

3CX quite effortlessly addresses all the major consideration points that an organization might have in choosing a phone system. Here are some of these points:

• There is no need for separate phone wiring to use 3CX VoIP Phone System for Windows. Since 3CX is installed on Windows, it can leverage your computer network to meet its infrastructure needs freeing you from extra cables and wirings. This can help save on cabling costs and unnecessary wiring clutter at the desktop.
• 3CX offers a simple web-based administration for installing and managing the VoIP phone system. This web based management portal makes working with 3CX a breeze. Say goodbye to expensive technician visits to add or change a user.
• If you have experienced a legacy PBX or key system before, you will know that the most tedious and costly part of owning a PBX can be the Moves, Adds and Changes or commonly known as MACs. With 3CX, you or your employees can move or switch offices at any time. No need for any re-wiring or re-configuration. 3CX Phone System for Windows just keep humming along seamlessly!
• Due to the extensive list of interoperable devices with 3CX, there is a tremendous amount of choice when it comes to selecting VoIP Phones, VoIP Gateways, Cordless Phones, Conference Phones, Servers, Headsets, etc. With so many brands and models to choose from, there is never the problem of vendor lock-in. Choose the hardware you like and it will work with 3CX. This really opens up tremendous options for clients.
• 3CX provides much needed flexibility and reliability in the VoIP system. With 3CX, you can use the powerful built-in VoIP capabilities to make and receive calls through your VoIP Provider of choice or you can use a standard VoIP gateway to make and receive calls through your regular telco lines. No more eggs in one basket!
• Since 3CX is a software based IP Phone System, there are significant cost savings over a proprietary hardware PBX. This makes the cost of acquisition and maintenance extremely affordable!

Cons of using 3CX Phone System

3CX is an excellent solution for most small and mid-sized organizations but it might not be for everyone. For example:

• Very small or extensively mobile businesses might still opt for a hosted phone system that gives them the convenience of a flat monthly fee with zero maintenance and support worries.
• Very large businesses may still go for a proprietary hardware-based solution, especially where there may be many widespread branch locations.
• Organizations that have in-house Linux development expertise may opt for an open source Asterisk based solution.

Conclusion:

Every organization’s needs may be different. When evaluating a phone system, it makes sense to define your specific requirements and budget before comparing vendors. Cost to purchase and ongoing cost to manage are also obviously important factors to consider. Ease of use is equally important.

To try out 3CX, you can download and evaluate it free of charge at:

http://www.3cx.com/phone-system/download-phone-system.html

To see 3CX and related products, check out the 3CX product page at the CI Gear VoIP Store:

http://www.cigear.com/3cx-phone-system-for-windows-c-159.html

Tagged in

Productivity Gains
Advanced features and integration between phone and computer enable users to be more productive and efficient. Popular features include Click To Call, Voicemail to Email, Find Me Follow Me, and Mobility.

Accessibility
Most systems allow a user to run an extension from a software phone that can run on a laptop or smart phone. This enables a user to take his/her extension along when traveling or working from home. There are tremendous savings to be found in mobile roaming and long distance charges.

Business Process Efficiencies
Many business process efficiencies can be realized by simply becoming more accessible. However, Presence and Intelligent Call Routing can help tremendously to further streamline business processes.

Web Integration
Connecting your web site to your phone can have a huge impact on sales and service. Wouldn’t you want to speak with your customer at the exact moment that they are interested in learning about your product?

Flexibility
With Business VoIP, there is no need to get locked into a contract for lines you may rarely use. Increasing or decreasing line capacity is a painless call or click away. You can also use virtual numbers to create a virtual local presence in another city. Be everywhere, all the time!

Moves and Changes
With simple web based administration, moves and changes no longer require a costly visit by a specialized technician. Also, when moving your business, VoIP makes it extremely simple and cost effective. No forklifts required!

The Aastralink RP phones were originally designed to work only with Microsoft Response Point phone systems including the Aastralink Pro RP system. Since the fate of Response Point is up in the air with Microsoft’s plans for further development on hold, some customers are looking for alternate solutions in case they outgrow RP or in case Microsoft decides to pull the plug on it.

This conversion release provides some investment protection for AastraLink RP customers since it helps simplify the upgrade path by preserving the customer’s investment in phones. This release will change the firmware in the Response Point phone to that of an Aastra standards-based SIP phone – allowing it to be used with standard based SIP PBXs such as the Aastralink Pro 160, 3CX, pbxnsip, Trixbox and hosted soft-switches, such as Asterisk, MetaSwitch and BroadSoft. The conversion process is easily accomplished through the phone’s Web UI, requiring only a TFTP server and the firmware downloaded from our website.

The new conversion firmware can be downloaded here.

Some benefits:

Check out our Skype Gateways page.

• Check in and Check out of guests
• Setting of guest extensions to Do not Disturb
• Blocking of external calls
• Schedule wake up calls
• Allow housekeeping to set room status
• Track guest calls and list them upon check out for easy billing

The 3CX hotel module can be integrated with many Hotel property management systems (PMS) such as Micros Fidelio, Roommaster, Hilton and more. Alternatively the receptionist can use the system directly through the provided web interface.

The Front Desk Phone Control Application is a simple to use windows or web based application that allows the front desk clerk to perform the above operations. This application can be used separately from the PMS system, and thus requires no integration at all. For many smaller hotels this can be good enough.

Billing

The hotel module will track calls made by a guest and output the list upon checkout for easy billing. The list of calls or the total of the calls can be outputted to screen or printer. In addition the information can be sent to the PMS system via CDR or SMDR records. For each case, the exact format can be customized.

Integration options

The 3CX Phone System hotel module can integrate with PMS systems in the following ways:

1. Via the 3CX Protocol, which resembles the Mitel SX 2000 hotel API, such that you can select the Mitel SX 2000 PBX in your hotel software.
2. Via the Fidelio protocol
3. Via middleware software called Lodging Link. Lodging link integrates with numerous hotel software systems and provides transparent integration with the 3CX PBX hotel module. Supported hotel software systems included can be found here: http://www.comtrol.com/aboutus/partners/hospitality/pms
4. Exchange of data can also be done on a custom basis. In this scenario, documentation must be provided by the reseller.

Some Property/Hotel Management Systems supported by 3CX

• Agilysys
• Autoclerk
• Booking Center MyPMS
• Cenium
• Maestro for Northwind
• Fidelity Hotel Management System
• Micros Opera
• Choice Hotels International
• Brilliant Hotelsoftware
• CMS Hospitality Guestcentrix
• Fidelio-Opera
• Micros-Fidelio
• Fidelio
• ResortSuite
• Gracesoft
• Hotec Expert Hotelier Suite
• Hotel Concept
• WebPMS
• Atrium INN client server systems
• HSS Hotel Software Systems
• Innfinity software systems
• InnRoad
• MyHMS innpoints
• Full list at http://www.comtrol.com/aboutus/partners/hospitality/pms

For more information on 3CX products, visit our 3CX products page or our 3CX information site

In yesterday’s installment of this series we discussed the internal router and other networing capabilities of the Atcom AG-188N (sold in North America by CIGear). The one thing that had me a bit puzzled was the implementation of VPN Tunneling. I’ve since been informed that the UDP method of VPN tunneling was implemented for a specific customer, and only works with that customer’s servers. However, the LT2P method does use standard L2TP, but without IPSec, which means it’s not a secure tunnel (there is no encryption used by default with L2TP). The most likely reason for using an L2TP tunnel would be to overcome NAT firewall issues, if for some reason you can’t use the IAX protocol and must use SIP instead. I’ve since added some information to yesterday’s article that includes a link to a blog post that shows how to setup an L2TP server on a Linux box. I’d still be interested in hearing from anyone that actually gets this working, since I haven’t really had the time to experiment with it yet.

Anyway, here are my final thoughts on this unit:

Let me state at the outset: I’m highly impressed with the AG-188N. Please keep that in mind as you read the paragraphs below, because reviews often tend to focus on areas where a product does not meet the reviewer’s expectations. In the case of the AG-188N, my expectations were met and for the most part, far exceeded. The one area where that was not the case was the documentation, and I’ll talk about that in a moment. But in my testing, this unit performed at least as well as any other VoIP adapter I’ve used, and I actually perceived the audio quality to be better than what I hear when using the venerable Linksys/Sipura adapters.

Atcom AG-188N

The Atcom AG-188N is a solid, well-built unit in an attractive charcoal-grey case. It’s very similar in size to the popular Linksys PAP2, and could be a drop-in replacement in any situation where only a single line (phone port) is required. What most impressed me during my testing was that everything I tested worked at it was supposed to, although not always exactly in the same manner as it would on a Linksys adapter (the need to press the star key to complete a 3-way call being one example). The device supports up to two SIP accounts, but in addition it supports one IAX account (something rarely found in a VoIP adapter in this price range), and you can also connect a PSTN line to the PSTN port on the back to use for calls to emergency services, local calls, etc. If a call comes in using any of the supported methods the phone will ring and you can talk, but call waiting does not appear to work “across technologies” (that is, if you’re on an IAX call and a call comes in on one of your SIP accounts, the device will report BUSY rather than giving you the call waiting beeps).

Call quality is excellent, although out of the box the incoming receiver volume was a bit low, but that was easily adjusted in the device’s configuration. I had no problems using this device with an Asterisk/FreePBX server, it worked as expected and was easy to configure (I daresay easier than the Linksys/Sipura adapters I’ve set up in the past). It does not have as many configuration options as the Sipura, but the settings that are missing are for the most part ones you’d probably never change from the defaults anyway. Actually, about the only settings I really wished for that I did not find were the ability to manually tweak certain tones (in particular, the tone that is sent when you leave the accidentally leave phone off the hook — this unit just provides a fast busy signal at roughly normal volume, which isn’t going to alert anyone that the phone is off the hook) but I’m probably the one person in 1,000 that would even care about that.

Honestly, I’m surprised that this unit hasn’t gained greater acceptance among VoIP users and service providers. Probably the greatest drawback of the unit is that many buyers will compare this with the Linksys PAP2 and notice that the PAP2 has two phone ports, while the AG-188N only has one. But on the other hand, the PAP2 does not support the IAX protocol, and does not include a router with DHCP support and a NAT firewall, not to mention QoS and VPN tunneling, if you can figure out how to configure the latter two. The Atcom AG-188N supports all of this, at a very reasonable price.

The documentation that is provided with this unit, to put it charitably, could be much better. No printed documentation is provided at all, but the included mini-CD contains documentation on several Atcom products including the AG-188N. Unfortunately, it’s in .doc format, which may not be readable if you don’t have Microsoft Word (although there are free online services that can convert .doc files to other formats — Zamzar comes to mind, but there are many others). You can go to the manufacturer’s web site to find PDF versions of the documentation, but at least in the case of the AT-188N, that documentation appears to be older than the .doc files on the disk. You may find that reading the previous articles in this series fills in some of the holes in the documentation.

Two specific issues with the documentation are that 1) it was apparently poorly translated from Chinese, with numerous spelling, punctuation, and syntax errors, not to mention being borderline incomprehensible in a few places (although it’s far from the worst translation I’ve ever seen), and 2) It is silent in some places where it should be more explanatory. For example, there’s an entire section of the manual on QoS (Quality-of-Service) configuration, but it’s titled “VLAN implement” (so many users would have difficulty finding it) and worse, it really doesn’t explain which of several possible configurations should be used in any given situation. For example, if you are using VoIP but you also have a computer (or a switch with multiple connected computers or devices) connected to the LAN port, how do you make sure that the VoIP packets get priority? That’s the sort of thing that needs to be explained in clear, easy-to-understand language.

And, there are some settings on which the manual is inexplicably silent. Worse yet, the entire section on VPN gives such sparse information that I sincerely doubt that anyone could figure out how to set up a VPN tunnel without getting additional clarification. It doesn’t help that they picked two of the most obscure VPN methods to support – one that’s used only by a single customer (though they don’t mention that in the manual) and the other an implementation that you rarely see anymore (L2TP with IPSec is somewhat common, though still not nearly as popular as some other methods, L2TP without IPSec much less so, although in a perverse way that may be a blessing in disguise – everything I’ve read about IPSec indicates it’s kind of a bear to install). I’m told that Atcom actually has a working OpenVPN implementation, but that it won’t fit in the flash memory and RAM of the device.

But really, none of this will matter to the typical user – there is a Quick Start Guide on the mini-CD that will be enough to get the average user going (or, you could refer to the previous articles in this series!) using IAX or SIP. Most of the router configuration on the AG-188N is fairly self-evident — if you’ve ever set up a router before, you shouldn’t have any difficulty with the one in this device. The places where things get somewhat complicated tend to be mostly places where typical users would fear to tread anyway.

If I could make a suggestion to Atcom, it would be this — For the next iteration of this device, please consider the following: 1) Enough memory for proper support of additional VPN tunneling methods, including OpenVPN, 2) More than one LAN port on the back of the unit (I know this will make it a bit larger, but you already have the AT-188N for those who need the smaller form factor) – it would be good to have at least four LAN ports, 3) the ability to select whether each individual LAN port is tunneled or not tunneled (when a VPN tunnel is active), 4) AT LEAST two phone ports, preferably four (if for no other reason than to make your device competitive with all the other two-line units), 5) Simpler QoS setup — build some default profiles for various situations, or better yet, let users assign QoS priorities on a per port basis (phone ports would default to high priority, LAN ports to medium priority, but each could be changed by the user with simple “radio button” selection), 6) A “greener” power supply — the “wall wart” you’re supplying with the AT-188N throws off too much waste heat (EDIT: And as noted, the “wall wart” that I received failed after approximately seven months’ continuous use), 7) And PLEASE, get someone who is a native English speaker to write or edit the English version of your manual (using proper grammar, spelling and punctuation!), and make sure you document ALL the settings on EVERY screen.

But even with whatever faults it may have, I think that any buyers of a single-line VoIP adapter (ATA) would be extremely happy with this device. It truly delivers more value than you’d expect for such a low price. When the worst thing you can say about a unit is that the documentation could be improved (and how many other products suffer from the same problem nowadays?), that’s really not much of a criticism.

One other thing that surprises me is that the sort of people who like to hack routers and/or network storage devices, and install rewritten and improved firmware, haven’t discovered this unit. When you consider all the energy that’s been put into modifying the firmware and then developing software packages for a device like the Linksys NSLU2, I’d have to think this device would be at least as attractive for that purpose. Sure, it doesn’t have any USB ports, but it’s a network connected device, which means that potentially it could store anything too large to fit in RAM on a network-attached storage device. Maybe the Atcom folks would not appreciate me mentioning the possibility that the firmware could be improved upon, but since the unit does have telnet access I have to think there’s a good chance it’s running some form of Linux “under the hood” — although I could be wrong about that — and I’m just a little bit shocked that those who enjoy digging into the internals of this type of device haven’t (yet) found the AG-188N attractive.

This is a device that really needs additional exposure among the VoIP community, which is one reason I wanted to review it. Sometimes it’s hard to get “word of mouth” going on a new device, but in my opinion this one deserves it. It’s a great, well-built device in any case, but with the ability to use IAX protocol and to tunnel its SIP connections via L2TP, you have two ways to handle those tough situations where most other adapters will connect to your server, and you can make their phone ring, but you only get one-way audio (or even no-way audio in some cases).

I’m going to close this article, and this series, with the list of features and specifications for this unit, from CIGear’s web site, but with the caveat that they probably got this from the manufacturer and I don’t believe it’s 100% accurate (just one example: Both this list and the AG-188N manual say that “Reverse polarity” is supported. But there’s no setting to turn it on or off, and I can tell you that it doesn’t appear to be on by default). So if a particular feature is really important to you, look back through the the screenshots in the previous articles of this series and if you don’t see that feature mentioned, there’s a chance it might not be implemented. This is just another example of a disconnect between the documentation and the actual unit.

Features

Enterprise, small office and residential applications

Key features:
Support two sip servers running at the same time.
Redundancy sip server support.
Support T.38 fax function
Support IAX2 protocol
NAT, Firewall.
DHCP client and server.
Support PPPoE, (used for ADSL, cable modem connecting).
Support major G7.xxx CODEC.
VAD,CNG.
G.165 compliant 32ms echo cancellation
E.164 dial plan and customized dial rules
Support Lifeline.
Hotline.
Call Forward, Call Transfer, 3-way conference calls
Call ID display
DND(Do Not Disturb),Black List,Limit List
Reverse polarity
Voice prompt
Increase Vlan and automatically upgrade configuration file encryption function
support IAX2 and FAX
Increase the time zone function

Data Features
Static/Dynamic WAN IP Addressing
PPPoE

Management
Web, telnet and keypad management.
Adjustable user password and super password
Upgrade firmware through HTTP, FTP or TFTP.
Telnet remote management.
Upload/download setting file
Auto-provision.
Safe mode provide reliability

Interface
Two RJ45 ports, one for WAN, one LAN.
One RJ11 PSTN port for lifeline
One RJ11 FXS port for analog phone.

Data Networking:

MAC Address
TCP:Transmission Control Protocol
DHCP:Dynamic Host Configuration Protocol
PPPoE：PPP Protocol over Ethernet
SNTP, Simple Network Time Protocol
STUN – Simple Traversal of User Datagram …
MD5 Message-Digest Algorithm
DNS：Domain Name Server
RTP: Real-time Transport Protocol
RTCP:Real-time Control Protocol
Telnet:Internet’s remote login protocol
HTTP:Hyper Text Transfer protocol
FTP:File Transfer protocol
TFTP:Trivial File Transfer Protocol

Call control /voip Features
SIP RFC3261,RFC 2543
Tone generation and Local DTMF re-generation according with ITU-T
G.711(A-law or u-law)
G729
AGC(Auto Gain Control)
G.168/165 compliant 16ms echo cancellation
AEC(Auto Echo Cancellation)
VAD (Voice Activity Detection)
CNG(Comfort Noise Generation

Enviromental

Electric requirements
Voltage: 9V ~ 24V
Power adapter: output DC 12V/450 mA

Operating requirement
Operation temperature: 0 to 40° C ( 32° to 104°F)
Storage temperature: -30° to 65° C (-22° to 149°F)
Humidity: 10 to 90% no dew

Regulatory compliance
CE, FCC part 15

Thank you to the folks at CIGear for their patience in answering my questions as I wrote this series of articles!

Disclosure: CIGear provided me with an Atcom AG-188N for review purposes, and allowed me to keep it after I was finished writing this series, and for that I am most grateful.

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Articles in the series: Review of Atcom AG-188N IAX+SIP ATA (VoIP adapter)

Part 1 – The unboxing
Part 2 – Initial setup using IAX
Part 3 – Setting the time and configuring outbound dialing
Part 4 – Setting up SIP, and securing the adapter
Part 5 – Networking and Internal Router
Part 6 – Final Thoughts and Summary Review
Part 7 – Addendum