Inter-Tel 5000 Network Communications Solutions
The Inter-Tel 5000 Network Communications solutions delivers VoIP communications, including connectivity, presence, collaboration, messaging, and other rich IP applications, including desktop video, and targets small- and mid-size companies. One innovative aspect of the Inter-Tel 5000 system is that it is perhaps the first IP-PBX system to utilize industrial-strength Compact Flash (CF) technology in its call processing units. This significantly increases reliability by eliminating spinning disks and other movable parts from the core system. The Inter-Tel 5000 is also unique in that it allows customers to leverage their existing digital endpoints, further enhancing customer value. Through the use of optional expansion units, customers can add up to 96 digital and analog endpoints, while still leveraging the benefits of VoIP.
Its presence management is one of its most powerful attributes. One of Inter-Telï¿½s presence tools is their Unified Communicator software, a communications management tool with powerful features such as routing rules, presence management options, personalized call handling, customized greeting, call history, consolidated contacts, and more. Another presence application is Connection Assistant, a workgroup tool that provides enhanced call handling, flexible programming of extensions, and screen-pops of frequently used applications. Another interesting presence tool is Telephony Manager, a Microsoft CRM tool integrated with Inter-Telï¿½s voice platforms via the OAI interface that allows you to initiate, manage, and track telephone conversations directly from Microsoft CRM; view contact and account information; and share information across departments.
Another unique feature of the Inter-Tel 5000 is its WAN failover solution to ensure continuity in the event of WAN or LAN disruption. In the event of a failure, the Inter-Tel 5000 still delivers its full feature set to users, in comparison to most competing products, which are reduced to just delivering dial tone. The 5000 is compatible with Inter-Telï¿½s Axxess platform and scalability is very good ï¿½ up to 63 systems can be transparently networked. Finally, the Inter-Tel 5000 supports several industry standards including SIP, MGCP, 802.11b, 802.3af (Power over Ethernet), TAPI, CT Connect, and more.
LignUp Communications Solution
Voice is becoming commoditized and the price of dial-tone is rapidly approaching ï¿½freeï¿½ with the help of companies such as Vonage and Skype leading the way. To survive, service providers must offer differentiated services and evolve from telephony providers to telephony integrated Application Service Providers (ASPs). LignUp enables service providers to offer hosted telephony services for small, medium, and distributed enterprises, as well as offer converged application services that integrate voice and presence into business applications and accelerate business processes.
LignUp Corporation delivers a powerful Web services-based VoIP communications platform deployed with hosted IP-PBX, IP-Centrex, Voice Mail, Unified Messaging, and IVR Web applications. The platform supports industry standard VoiceXML, but it also supports LignUpï¿½s proprietary Media Control XML (MCTRL), and LignUp Call Control XML (CCTRL) languages. These are more powerful XML-based languages that Lignup has developed since the VoiceXML spec is still immature and incomplete. Using MCTRL or CCTRL developers can create custom voice solutions, voice-enable enterprise applications, voice-accelerate business processes, and more. LignUp Web Services enable Web developers to easily access powerful call control and media processing capabilities via XML. In fact, LignUpï¿½s hosted PBX, VoiceMail, Unified Messaging, and other applications have been built using LignUp Web Services.
Essentially, LignUp allows service providers to launch and operate VoIP services with lower capital expenses, as well as allowing these players to scale operations as the business scales. One key aspect of the LignUp solution is that it offers a comprehensive solution as opposed to their competitors which only offer pieces of the solution. This eliminates the costs to integrate solutions from a variety of vendors.
Another key advantage of the LignUp solution is that it is uses open standards including ï¿½nativeï¿½ SIP with proven interoperability with a variety of SIP devices. Also, since it supports SIP, the solution can connect directly to VoIP networks such as Level3. Scalability and reliability are excellent with their modular, clustered architecture. Each LignUp component ï¿½ LignUp Call Directors for call control, LignUp Media Servers for media processing, LignUp Web Applications (Hosted PBX, VoiceMail, Unified Messaging, IVR and others) ï¿½ can be deployed on a single server or distributed on separate servers. In addition, each component can be clustered separately for more flexible deployments. Redundancy is implemented simply by deploying two of each component. If any one server fails, the other continues providing the function.
The traditional process of telephony application development would require the work of scarce and expensive CTI developers, working on proprietary or platform specific TAPI, or other APIs. Applications would have to be customized to each platform and only one or two significant applications would be able to get to market per year. According to Lignup, ï¿½These challenges raise significant barriers to entry for most aspiring service providers, and made even large providers (all of whom are margin sensitive) tentative to embark on fully pursuing the development of high-value applications.ï¿½ Furthermore Lignup stated, ï¿½Developers do not need to understand the details of SIP, and can use familiar Web development tools such as IBM Websphere, BEA Weblogic, MSN.net, JSP, ASP, Perl, CGI, etc. This enables service providers to deliver dozens of innovative applications per month.ï¿½ TMC Labs was impressed with the comprehensive LignUp Communications Platform, which allows for rapid application development and deployment thus enabling service providers to quickly incur incoming revenue.
DashPhone CXP is a DAS module that makes applications written for Cisco 7940/7960/7970 series of IP phones available on any IP phone and IP endpoint that is supported by DAS. CXP can sit between any Cisco XML application server and any supported IP phone or IP endpoint, and deliver the application transparently to the device. Millenigence claims that DashPhone CXP is the first and only product available in the market that delivers Cisco XML IP phone applications to non-Cisco IP phones.
Using CXP, various phones such as Avaya 4620 or Siemens optiPoint 600 can take advantage of a host of Cisco XML applications that are already developed by software vendors for Cisco IP phones. Since Cisco was the first company to popularize a data interface for IP phones, many more applications are available for IP phones made by Cisco compared with any other vendor. CXP makes these applications available on other IP endpoints.
DashPhone CXP allows data-centric applications and services written for Cisco IP phones to not only run transparently on non-Cisco IP phones, but also PDAs, and Wi-Fi or WAP enabled cell phones with zero programming. An employee can use his or her IP phone, PDA, or cell phone as a terminal to interact with the same enterprise data. Essentially, CXP is middleware that sits between a Cisco application server and non-Cisco phones to translate the XML pages intended for a Cisco IP phone into whatever format the non-Cisco phone is able to understand, display, and respond to. Dashboard CXP is fully J2EE based and therefore capable of running on virtually any operating system and server environment.
Using DashPhone CXP, popular Cisco XML applications (which was just intended for Cisco gear) can now be run on a vast range of IP phones. Developers can easily use Cisco XML for writing their applications and need not worry about the target device.
Thus, if you are developing or have developed Cisco specific applications and services for the 7940 and 7960 IP phones, CXP allows your applications to reach new markets unmodified, without any re-coding. Further, with CXP, you can even run and test the application on your PC browser, making development much more convenient.
When you think of Nortel you think of ï¿½big ironï¿½ PBXs: reliable and extensible through many third-party applications, targeting medium-to-large businesses ï¿½ but you certainly donï¿½t associate Nortel with the SOHO (small office/home office) or SMB market, now do you? Well, times have changed. Nortel is targeting their new Business Communications Manager 50 (BCM50) product directly at small single sites, franchises, or branch offices. The Business Communications Manager 50 platform, which just launched in May, is designed for 20 stations with room to grow to 40+ stations. This 20ï¿½40 station size range is certainly a new benchmark for Nortel. We bet if you called Nortel five years ago for a 20ï¿½40 station phone system theyï¿½d probably laugh at you. Unlike larger businesses which tend to stay with their existing phone system hardware for a 10ï¿½20 year lifecycle, small-to-medium businesses are always looking for a competitive advantage and tend to swap out their phone system hardware much more quickly. Further, mi llions of new small businesses are created each year requiring new phone systems to be purchased.
Well, the Business Communications Manager 50 platform is ideal for businesses that require advanced capabilities, such as robust telephony features, voice messaging and unified messaging, IP networking, Internet/intranet access, skills-based routing, IP telephony to usersï¿½ desktops, and an integrated router option for Ethernet or ADSL broadband access.
According to Nortel, ï¿½Our goal is to make ï¿½convergenceï¿½ affordable and available to the smallest business sites.ï¿½ They also stated, ï¿½Through open standards and an ï¿½evergreenï¿½ development strategy, Business Communications Manager 50 platforms fit well in hybrid environments that contain a mix of analog, digital, or IP services. And since it interworks with other Nortel key/PBX systems, larger Business Communications Manager systems, and with our portfolio of convergence call servers, you have a smooth migration path.ï¿½
Nortel told us, ï¿½The Nortel Business Communications Manager 50 system combines the best elements of high-end digital PBX phone systems, cutting-edge convergence solutions and robust data networking in one affordable package. By integrating advanced data networking and comprehensive telephony features in a single device, Business Communications Manager 50 delivers a level of system integration and flexibility rarely seen in the industry ï¿½ and certainly uncommon for small business locations.ï¿½
Installation and configuration are very easy to do using their included software, which is important in small-to-medium businesses that often do not have their own dedicated telecom department and instead rely on their IT staff to configure various phone system parameters. One innovative aspect of the system, which Nortel has been doing for years in their other phone products, is that you can program functions using any connected telephone set. One final innovative feature of note is that the BCM50 sports more than 200 telephony features which you usually only see in high-end PBXs.
We all know the story of Pandoraï¿½s Box which unleashed evil upon the world. Well, Pandora Networkï¿½s appliance ï¿½boxï¿½ unleashes a converged platform that includes IP Centrex, ACD, Web Contact Center, video, Instant Messaging services (AOL, Yahoo, MSN compatible), and collaboration. We certainly like their corporate name but we took pause when reading their application for this award when we read that the product name was called Worksmart. A converged VoIP platform was certainly not the first thing to come to mind when we read ï¿½Worksmart.ï¿½ Our ï¿½critiquingï¿½ of their product name aside, this is one very impressive product.
First off, Worksmart is a plug-and-play IP-enabled enterprise communications appliance that features IP telephony, call center functionality, and more. One obvious benefit of this converged solution is that instead of purchasing and integrating multiple products from many vendors, Pandora Networks packs them all into an easy to use and operate appliance. Using a single Worksmart appliance means you donï¿½t have to purchase an IP-PBX, video conferencing server, instant messaging server, collaboration server, and an online Web contact server separately. It connects to either your traditional carrier or to Pandoraï¿½s own SIP-powered IP telephone network that allows inbound and outbound phone service. One unique aspect of this solution is that unlike IP Centrex services, Pandora can deploy their on-demand services from their Central Offices or allow the customer to operate it on their premise and to blend with TDM interfaces when desired.
Worksmart enables your employees to engage in voice, video, and collaborative conversations from anywhere, using one simple desktop application. According to Pandora, cost savings are up to 80 percent less than single-point solutions. Pandora Networks stated, ï¿½Thereï¿½s no complex administration, no third-party hardware integration, no reconfiguring your network, no weeks lost on integration and training ï¿½ just plug it in and turn it on. The powerful Web-based management interface allows you to start working in just a few minutes. Users will love having just one desktop application that provides all of their communication needs and allows them to use any phone as an extension ï¿½ including a cell phone.ï¿½
RingCentral provides customers with a personal toll-free and/or local area telephone number with an integrated auto-attendant that can screen and forward calls and take messages and faxes ï¿½ all with no customer hardware or software required. When callers phone or fax a userï¿½s RingCentral number, RingCentral automatically handles the call routing. You have the option to route the call to a single number or simultaneously to multiple numbers where you can be reached, or you can send the call to voice mail. You can also define schedule preferences for when certain rules are in effect, such as only dial your cell phone during the evening.
If you are online, a pop-up screen notifies you of the call that includes caller audio preview and CallerID. Users of RingCentral can screen who is calling and see their caller ID before deciding whether to accept the call, send it to voice mail, or reject it outright. They can also direct the call to any phone including their cell phone, land line or VoIP phone or they can direct the call to voice mail. For missed or ignored calls, RingCentral can notify users of their messages by phone, pager, Web, or e-mail.
RingCentral also includes full-featured, computer-based faxing capability. RingCentral delivers incoming faxes as e-mail attachments. It can also store them on RingCentralï¿½s servers for Web-based access. Users can also send faxes from within any Windows application.
Using this service you have the capability to manage messages, faxes, and call records over the Web, and over the phone, and on a PC using their software. The software includes detailed logs of all incoming and outgoing calls, and it presents calls, messages, and faxes in a Web-based user portal. The service comes with a unique Windows Call Controller application that can be used to manage calls, messages, and call logs locally on a userï¿½s PC. Thereï¿½s also an online address book that integrates with Outlook and Outlook Express and can be used for caller identification and easy dialing.
A business plan offered by RingCentral grants you a virtual PBX with auto attendant and extensions for individual employees, who get personalized dashboards to manage inbound calls. Each extension can be programmed with its individual call-forwarding rule. RingCentral also supports direct dial and direct fax numbers, making it possible for a person or department to be reached through both the main company number via an extension, as well as by a direct phone number.
One unique feature of RingCentralï¿½s Call Controller is the option of typing a quick message that goes through a text-to-speech engine; the message is immediately played back to the caller as a customized greeting, composed on-the-fly. This is useful when on another line or unable to take a call. The custom message that users compose can be based on their availability and the CallerID of the caller, for example: ï¿½Hi, John, Iï¿½m on an important call, but Iï¿½ll call you back in 15 minutes.ï¿½ One final innovative feature is the integrated click-to-call RingOut function. This feature allows you to quickly call-back numbers stored in call logs, voice mail, and directories.
SER Solutions, Inc.
TSP500 Outbound Dialer
SERï¿½s TSP500 is not your ordinary predictive dialing switch. Sure, it has full compliance with all state and federal do not call regulations. Sure, it has excellent scalability supporting up to 384 agents and sure, it leverages SERï¿½s SmartPace VI dialing algorithm, which SER claims generates more connects as compared to other predictive dialers. But the real innovation is that the TSP500 enables contact centers to consolidate their operations through the use of integrated VoIP. Their VoIP support allows remote agents located either in the U.S. or abroad to connect to the predictive dialer switch using H.323 or SIP. The switch also delivers enhanced Caller ID functionality that allows contact center operators to display both the telephone number and designated name of the caller or client for effective campaign management and compliance with FCC and FTC mandates.
The TSP500 reduces total ownership and communications costs by allowing contact centers to utilize VoIP to distribute information, agents, and technology across multiple centers. According to SER, ï¿½We believe the TSP500 is the first predictive dialing switch that can seamlessly scale to 384 agents, provide native support for VoIP connectivity, enhanced Caller ID capabilities, and generate more connects than other predictive dialers-all on a single system.ï¿½
The TSP500 ensures peak agent efficiency by leveraging SERï¿½s SmartPace VI dialing algorithm which maximizes live connects while eliminating unwanted busy signals, answering machines, fax machines, and ring no answers. SERï¿½s voice recognition detects a human voice within milliseconds, as well as busy signals, ring no answers, and telephone company SIT tri-tone intercepts. This feature boasts productivity and ensures a natural call flow by allowing agents to hear the first ï¿½helloï¿½ instead of a silent pause.
Since the TSP500 supports the industry standard SIP protocol and is interoperable with several SIP proxy solutions, the TSP easily supports work at home agents through VoIP and SIP without requiring the agents to use a soft phone or IP phone solution. Remote sites can utilize a VoIP gateway or ATA (analog telephony adaptor), which converts the IP audio stream back to an analog TDM stream, with agents connected through simple analog phones. The gateways can communicate with each other using either the H.323 or the SIP protocol. In this configuration, as long as the gateways at the remote site have power, the conversations will not be interrupted, even if the desktops lose power.
The other possible configuration is to connect the TSP500 directly to either IP phones or an IP soft phone on the agentï¿½s desktop computer using SIP. While this can be used when the remote agents are located together, this provides the greatest value with agents that are dispersed, because no remote gateway is required. This configuration works with several commercially available off-the-shelf SIP proxy servers.
Toshiba America Information Systems, Digital Solutions Division
Toshiba Strata CIX
Toshiba is well known for its relatively inexpensive, feature-rich phone systems targeting the SMB market. Toshibaï¿½s Strata CIX is no exception as it is designed for small- to medium-sized enterprises or larger corporate users with multiple sites with support up to 672 ports. It can be run as a pure IP system or can be TDM enabled, allowing enterprises to migrate to IP at their own pace.
One of the most innovative aspects of the Toshiba CIX is its support for several advanced applications on a single system utilizing the embedded Strata Media Application Server (MAS). MAS is one of the first devices to combine voice applications from multiple vendors onto a single platform using Intelï¿½s Host-based Media Processing (HMP) technology. This lowers the entry cost to many applications for the small to medium enterprises because it eliminates the need for multiple hardware platforms to support each application separately. MAS applications include Auto Attendant, Voice Mail, Automated Speech Recognition, Text to Speech, Unified Messaging, Interactive Voice Response, Automatic Call Distribution and Reporting, Web-based Personal and System Administration, Web-based Telephone Applications, ACD/MIS, and other third-party applications. If there is an application not included, you can build it yourself using an easy to use script editor that works with MAS to interpret code, process functions, follow c ustom routing, and more.
Strata CIX supports the SIP standard and it has been designed to deliver virtually every feature to every user, regardless of the type of device they are using, whether they are static or mobile. The system supports IP phones, IP wireless handsets, both analog and digital telephones, IP softphones on laptops, and tablet PCs. Of course, Toshiba hasnï¿½t forgotten their TDM roots so they support both pure IP and TDM. This allows users to choose how they will maximize their systems and migrate existing equipment. One final feature of note is Toshibaï¿½s new personal administration tool, ï¿½My Phone Manager,ï¿½ which enables individual users to easily program the LCD on the telephone and program speed dial buttons and feature buttons via their PCï¿½s Web browser. This helps alleviate administrator support.
SmartOnline Expandable Rack/Tower UPS System
Released back in November, Tripp Liteï¿½s SmartOnline Expandable Rack/Tower UPS battery protection system is perfectly suited for mission-critical VoIP equipment. In fact, this is no ordinary battery-backup system. This product is the first to be certified by Cisco Systems as capable of shutting down connected Call Manager servers in VoIP applications.
SmartOnline features a compact tower/rackmount housing and protects systems from the economic damage associated with the disruption and downtime caused by power blackouts, voltage fluctuations, and transient surges. Tripp Lite claims a continuous sine wave output with zero transfer time. In the event of extreme power events, the unit transfers the equipment load to battery power until stable power is restored. If the power disruption outlasts the systemï¿½s battery (configurable) capacity, connected applications and operating systems are signaled to shutdown in an orderly manner before all power is lost. This includes the popular Cisco Call Manager VoIP application.
This product offers a unique combination of runtime scalability, simple plug-in installation, outlet customization, and hot-swap serviceability for both power electronics and battery systems. The battery system and receptacle panels are modular, making this unit ideal for later expansion as budgets or application-related hardware needs evolve. In the event of a maintenance need, systems can remain connected and available during service, ensuring application uptime.
Another distinctive feature is the informative alphanumeric LCD front-panel display. Even many network managers need assistance in understanding their operating conditions. With power being such a critical component, there should be no doubt. The LCD display greatly enhances all usersï¿½ understanding of operating conditions, unlike traditional LED indicators. Finally, Tripp Lite backs up their products with $250,000 worth of Ultimate Lifetime Insurance.
The Vegastream 400 supports both SIP and H.323, has very good scalability and most importantly it has excellent voice quality with minimal latency. TMC Labs knows this since we recently tested the Vega 400 in the labs. The Vega 400 supports four E1/T1 interfaces and can be configured to support one to 120 VoIP channels (four E1s).
The Vega 400 has a very distinguishing feature that differentiates it from many of its competitors. That is, many competing gateways can also do 120 channels, but only using the G.711 codec and not other codecs. On the other hand, the Vega 400 can do 120 channels using G.711 but also 120 channels using, say G.729 or G.723.1. Due to the DSP or processor requirements many competing VoIP gateways can only do, for example, 90 out of 120 channels using the G.729 codec. Similarly, it supports T.38 faxing and it supports the full 30/24 channels per E1/T1 of fax simultaneously where as some other gateways only do a few.
Another unique feature is that the DSPs are on PCMCIA cards, which slide in the back of the chassis. Upgrades (i.e., from one T1 to four T1s) can be done without opening the unit. This flexibility allows you to start with a low-density gateway at a low cost, and add capacity as needed. One final feature of note is that when you first boot the Vega 400 you can select either a SIP or H.323 stack, which is a really nice feature. Many other solutions require that you manually download a separate firmware to install a different VoIP stack.
eQuality ContactStore for IP Solution
ContactStore for IP captures voice conversations and corresponding computer desktop screen activities in VoIP environments. Providing converged voice/data networks with a sophisticated and robust feature set, it includes the functionality of ContactStore, and enables large organizations and small- to medium-sized businesses (SMBs) to record VoIP calls while simultaneously meeting industry compliance requirements.
VoIP is fast becoming a mainstream technology for contact centers. Further, because of the shift toward remote agents and virtual contact centers and the move to centralizing systems like call recording, the ability to record VoIP calls becomes imperative. Witness Systemsï¿½ ContactStore for IP solution utilizes the Cisco IP telephony infrastructure to capture valuable business intelligence ï¿½ in the form of both customer and competitive insight ï¿½ by simply recording, evaluating, and analyzing customer interactions.
Witness Systems claims that ContactStore for IP was the first ï¿½software onlyï¿½ solution introduced to the market and they claim more VoIP recording deployments (650) than any competitor. ContactStore for IP leverages a Web-based architecture that scales from a single seat system to a distributed multi-site enterprise with thousands of channels, providing a single view of all customer contacts.
The product features the ability to ï¿½record all callsï¿½ as well as ï¿½record on demand.ï¿½ One unique feature is that it performs stereo recording to optimize the clarity of recorded speech and to aid in automated conversation analysis. This enables both sides to be recorded separately, which is important for dispute management. Also, word spotting and speaking authentication products can be used to only analyze either the caller, the agent, or both depending on your needs.
Each of the recorded interactions are assigned contact attributes, making it easy to retrieve, review, report, and analyze customer experiences for compliance requirements and quality purposes. Another interesting feature is ContactStore IP E-mail Call, which lets you e-mail the entire interaction at any point during call. The system can also be set up to trigger a recording based upon a CTI event. Another nice feature is that you can press the record button on the IP phone at any point during the call. An agent can capture and save the entire interaction and not just the portion from where the agent pushed the button. Also, you can start/stop and pause a recording to omit confidential information such as security passwords.
Xten Networks, Inc.
Xten Pocket PC SIP Softphone
Xten is well renowned for their SIP softphone clients, which typically have excellent voice quality with good codec support, and a minimal software footprint. Xtenï¿½s Pocket PC SIP Softphone is no exception. It supports open standards, including SIP obviously, but also optional G.729a, wideband codecs, AEC, VAD, and a superior audio mixer provide for excellent call quality and user comfort. According to Xten, ï¿½This product can be combined with open source open standards PBX offerings not unlike Asterisk to offer a wireless PBX solution. Compliant with Sylantro and Broadsoft this software is a good fit for IP Centrex as well.ï¿½ Everyone talks about when Skype will be embedded onto smart phones. Well, Xtenï¿½s Pocket PC SIP softphone client today can run on the new Windows Mobile 2005 operating system allowing you to connect to any SIP-based ITSP or SIP gateway.
Xten has over 1 million endpoints deployed and that number is growing rapidly. They also offer training for those who implement their SDK. Xten told us, ï¿½We listen to our customers and closely monitor the market, we have people in the IETF that monitor industry movement and give us guidance as to where we should be focusing our efforts, this really helps us get a head start on any perceived competition.ï¿½
The Pocket PC-based softphone supports Automated Echo Cancellation (AEC) thus negating the need for a headset. You can simply use your Pocket PC device just like a phone. The softphone client supports all the various call controls including transfer, blind and supervised, as well as three-way conferencing.
MX30 Enterprise Media Exchange
Zultsys is an interesting affordable ï¿½convergedï¿½ solution targeting the SMB (up to 30 users). The MX30 integrates voice, data, video, and fax, and provides the functions of an IP PBX, firewall, NAT, and VPN ï¿½ a complete solution for a small business or branch office, all in a single CPE appliance. The MX30 is designed specifically to connect businesses to Internet Telephony Service Providers (ITSPs) using SIP. The MX30 utilizes any broadband connection for its trunking to receive and route all calls to destinations external to the company. This allows customers to realize the full benefits of VoIP and save money through the lower rates offered by ITSPs, such as Level3, Deltathree, etc. Analog and ISDN basic rate connections are only provided as a backup in case the ITSP connection fails.
According to Zultys, ï¿½The MX30 follows the MX1200 and MX250 in providing a completely open standards platform for integrating the functions of an IP PBX, Internet gateway, network server, and application server. The MX30 is not a ï¿½stripped downï¿½ version of its larger siblings, but contains all of the features and functionality a smaller business will need in order to deploy a modern VoIP communications system.ï¿½
The MX30 is capable of networking with multiple MX systems over a WAN. This provides toll bypass savings and allows a small business to preserve its investment as it expands. Zultysï¿½ MXgroup software allows companies to network all sites, creating a single communications platform, enabling telephony, video, fax, voice mail, instant messaging, and presence, regardless of location.
The MX30 is unique in combining the features of an IP-PBX, fax server, application server, VPN, as well as connections to ITSPs, all in a single box. Further, their target market of 30 seats and under brings a new level of advanced VoIP functionality to the SMB market. Finally, every Zultys product including the MX30 is based entirely on open standards, such as SIP and VXML ensuring you are not locked into a single vendor when it comes to IP phones or VXML applications such as IVR, voice mail, etc. IT
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