IP Gateways

VocalNet's line of IP Gateways are designed for easy VoIP deployments within existing network architectures, allowing our customers to enjoy the benefits of VoIP today. These access devices are adaptors that connect conventional telephones and legacy telephony systems to our VoIP network, permitting high-quality audio communications over IP. Leverage our hardware resources today to enable cost-effective converged voice and data solutions for your business!

Select a manufacturer below to view our preferred IP Gateways, or contact a VocalNet sales representative to engineer the most appropriate solution for your individual needs.

ADTRAN

Total Access 900/900e Series VoIP Business Gateways

The Total Access 900 Series of IP Business Gateways combine the functionality of ADTRAN's industry-leading integrated access devices with a SIP and analog gateway to provide Incumbent Local Exchange Carriers (ILECs), Competitive Local Exchange Carriers (CLECs), and Internet Service Providers (ISPs) a cost-effective IP network strategy for VoIP deployment, with support for legacy equipment. The Total Access 900 and 900e Series allow carriers to deliver SIP trunks, hosted PBX, and other voice and data services such as Dedicated Internet Access (DIA) to small and medium businesses, quickly and cost-effectively.

Total Access 904

Seamless voice and data integration over VoIP based network architectures
WAN: Single T1 interface
Single DSX-1 for PRI/T1 CAS handoff to PBX
4 analog FXS POTS interfaces
Single 10/100Base-T interface
Includes SIP gateway, robust IP router, stateful inspection firewall with Denial of Service prevention
Compatible with industry leading soft switches and call agents
Dynamic bandwidth allocation
On board G.729 and G.711 CODEC support for voice compression
G.168 Echo Cancellation
Low latency, WFQ, CBWFQ Quality of Service (QoS)
Command Line Interface (CLI), Web-Graphical User Interface (GUI)
Network Address Translation (NAT) for IP Address Concealment
AC power with battery backup
Feature-rich ADTRAN Operating System (AOS)
Routing protocols: static, RIP, OSPF, BGP

Total Access 908

Seamless voice and data integration over VoIP based network architectures
WAN: Single T1 interface (908), Quad T1 interfaces, two can be used for customer facing PRI/T1 (908e)
Single DSX-1 for PRI/T1 CAS handoff to PBX (908)
8 analog FXS POTS interfaces
Single 10/100Base-T interface (908), Dual 10/100Base-T interfaces (908e) for WAN or DMZ applications
Includes SIP gateway, robust IP router, stateful inspection firewall with Denial of Service Prevention
Compatible with industry leading soft switches and call agents
Dynamic bandwidth allocation
On board G.729 and G.711 CODEC support for voice compression
G.168 Echo Cancellation
Low latency, WFQ, CBWFQ Quality of Service (QoS)
Command Line Interface (CLI), Web-Graphical User Interface (GUI)
Network Address Translation (NAT) for IP Address Concealment
AC power with battery backup
Feature-rich ADTRAN Operating System (AOS)
Routing protocols: static, RIP, OSPF, BGP
908e features T1 bonding with MLPPP/MLFR for high-speed network connections and optional hardware encrypted DES/3DES/AES IPSec VPN

Total Access 908e

Seamless voice and data integration over VoIP based network architectures
WAN: Single T1 interface (908), Quad T1 interfaces, two can be used for customer facing PRI/T1 (908e)
Single DSX-1 for PRI/T1 CAS handoff to PBX (908)
8 analog FXS POTS interfaces
Single 10/100Base-T interface (908), Dual 10/100Base-T interfaces (908e) for WAN or DMZ applications
Includes SIP gateway, robust IP router, stateful inspection firewall with Denial of Service Prevention
Compatible with industry leading soft switches and call agents
Dynamic bandwidth allocation
On board G.729 and G.711 CODEC support for voice compression
G.168 Echo Cancellation
Low latency, WFQ, CBWFQ Quality of Service (QoS)
Command Line Interface (CLI), Web-Graphical User Interface (GUI)
Network Address Translation (NAT) for IP Address Concealment
AC power with battery backup
Feature-rich ADTRAN Operating System (AOS)
Routing protocols: static, RIP, OSPF, BGP
908e features T1 bonding with MLPPP/MLFR for high-speed network connections and optional hardware encrypted DES/3DES/AES IPSec VPN

Total Access 912

Seamless voice and data integration over VoIP based network architectures
Up to 12 analog POTS interfaces
DSX-1 for PBX connectivity
Compatible with industry leading soft switches and call agents
Dynamic bandwidth allocation enables more efficient bandwidth utilization
Standardized G.729a voice compression requires less bandwidth per voice call
Integral full-featured IP router for data support and Internet access
Stateful inspection firewall for network security
Quality of Service (QoS) for delay sensitive traffic like VoIP
Command Line Interface (CLI) mimics industry de facto standard
Network Address Translation (NAT) for IP Address Concealment
Feature-rich ADTRAN Operating System (AOS)

Total Access 916

Seamless voice and data integration over VoIP based network architectures
Up to 16 analog POTS interfaces
DSX-1 for PBX connectivity
Compatible with industry leading soft switches and call agents
Dynamic bandwidth allocation enables more efficient bandwidth utilization
Standardized G.729a voice compression requires less bandwidth per voice call
Integral full-featured IP router for data support and Internet access
Stateful inspection firewall for network security
Quality of Service (QoS) for delay sensitive traffic like VoIP
Command Line Interface (CLI) mimics industry de facto standard
Network Address Translation (NAT) for IP Address Concealment
Feature-rich ADTRAN Operating System (AOS)

Total Access 916e

Seamless voice and data integration over VoIP based network architectures
Up to 16 analog POTS interfaces
DSX-1 for PBX connectivity
Compatible with industry leading soft switches and call agents
Dynamic bandwidth allocation enables more efficient bandwidth utilization
Standardized G.729a voice compression requires less bandwidth per voice call
Integral full-featured IP router for data support and Internet access
Stateful inspection firewall for network security
Quality of Service (QoS) for delay sensitive traffic like VoIP
Command Line Interface (CLI) mimics industry de facto standard
Network Address Translation (NAT) for IP Address Concealment
Feature-rich ADTRAN Operating System (AOS)

Total Access 924

Seamless voice and data integration over VoIP based network architectures
Up to 24 analog POTS interfaces
DSX-1 for PBX connectivity
Compatible with industry leading soft switches and call agents
Dynamic bandwidth allocation enables more efficient bandwidth utilization
Standardized G.729a voice compression requires less bandwidth per voice call
Integral full-featured IP router for data support and Internet access
Stateful inspection firewall for network security
Quality of Service (QoS) for delay sensitive traffic like VoIP
Command Line Interface (CLI) mimics industry de facto standard
Network Address Translation (NAT) for IP Address Concealment
Feature-rich ADTRAN Operating System (AOS)

Total Access 924e

Seamless voice and data integration over VoIP based network architectures
Up to 24 analog POTS interfaces
DSX-1 for PBX connectivity
Compatible with industry leading soft switches and call agents
Dynamic bandwidth allocation enables more efficient bandwidth utilization
Standardized G.729a voice compression requires less bandwidth per voice call
Integral full-featured IP router for data support and Internet access
Stateful inspection firewall for network security
Quality of Service (QoS) for delay sensitive traffic like VoIP
Command Line Interface (CLI) mimics industry de facto standard
Network Address Translation (NAT) for IP Address Concealment
Feature-rich ADTRAN Operating System (AOS)

Grandstream

GXW IP Analog Gateway Series

The GXW IP Analog Gateway Series enables small and medium businesses to create seamless office environments, integrate traditional phone systems into a VoIP network and efficiently manage communication costs. The GXW Series is designed for full interoperability with leading IP-PBXs, Softswitches and most SIP-based environments and offers 4 or 8 port models and a video surveillance port on the GXW410x models.

The GXW410x FXO Series enables businesses to seamlessly connect multiple locations (up to 8 PSTN lines per location) to an IPPBX system, or an existing traditional phone system. The GXW-410x features 4 or 8-port FXO interfaces, dual 10M/100M network ports with integrated router, up to 3 SIP account profiles, 1 or 2-stage dialing, caller ID, T.38 fax, and programmable PSTN line settings for various different countries/regions. In addition, the GXW410x supports a comprehensive list of voice codecs including G.723.1, G.711(a/u-law), G.729A/B, G.726, GSM, and iLBC (pending), and H.264 video codec (up to 30fps and CIF resolution) for the video surveillance port.

The GXW40xx FXS Series has a compact and quiet design (no fans) and offers superb audio quality, rich feature functionality, strong security protection, good manageability, and compelling price-performance ratio. It is auto-configurable, remotely manageable and scalable. It features a 4 8, or 24-port FXS interface for analog telephones, dual 10M/100M network ports with integrated router, PSTN life line in case of power failure, and an RS232 serial port for administration. In addition, it supports 2 SIP account profiles, caller ID for various countries/regions, T.38 fax, flexible dialing plans (pending), security protection (SIPS/TLS), comprehensive voice codecs including G.723.1, G.711 (a/u-law), G.726, G.729A/B/E and iLBC.

Digium

Digium Wildcard TDM400P

Digium Wildcard TDM400P

The Wildcard TDM400P is a half-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC.

Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system.

The TDM400P takes the place of an expensive channel bank and brings the system price point to a low level. By using S110M and X100M modules with the TDM400P, one can create a solution with support for a range of telephones. To scale this solution, simply add additional TDM400P cards populated with modules.

Note: The 12V power connector is required for the operation of FXS modules. It is not required for the operation of FXO modules.

Boxed TDM Retail Package Includes

TDM400P
FXS and/or FXO Modules, depending on which bundle was ordered
Hardware Installation manual
RJ11 cables (one for each active port)
Digium | Asterisk reversible screwdriver
Digium | Asterisk mouse pad

Digium Wildcard TDM2400P

Digium Wildcard TDM2400P

The Wildcard TDM2400P is a full-length PCI 2.2-compliant card for connecting analog telephones and analog POTS lines through a PC. It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines.

Using Digium's Asterisk Open Source PBX software and standard PC hardware, one can create a telephony environment that includes all the sophisticated features of a high-end business telephone system.

Using an industry-standard bursting, bus-mastering PCI interface chip that is found within millions of PC systems worldwide, and Digium's patent-pending VoiceBus technology, the TDM2400P eliminates the requirement for separate channel bank and T1 interface cards, at an industry-leading price. The quad-FXO and quad-FXS modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 256 taps of echo cancellation for superior echo cancellation on both FXO and FXS interfaces. Scaling of this solution is accomplished by adding additional TDM2400P cards.

Note: The 12V power connector is required for the operation of FXS modules. It is not required for the operation of FXO modules.

Quintum Gateways

The Tenor VoIP MultiPath Switches Guarantee Voice Quality on VoIP Calls
Quintum's Tenor VoIP solution is extremely flexible and scalable, making it ideal for the wide range of environments and applications that exist in enterprises today. The Tenor provides full compliance with both H.323 and SIP IP telephony standards, ensuring its interoperability with leading third-party hardware and software. Including unique technology such as SelectNetT, PacketSaverT, MultiPath Call Routing and NATAccessT, Tenors offer:

Tenor

Tenor AX Series

A more intelligent way to implement VoIP

The Tenor AX Series offers:
. 8, 16, 24 or 48 Analog line and trunk interfaces
. Up to 48 simultaneous VoIP calls
. Available in MultiPath or Gateway configurations
. Support for external Quintum Call Routing Server*
. Support for external Quintum Tenor Monitor
. Support for external Remote Management Session Server*
. Transparent MultiPath Call Routing
. Answer and Disconnect Supervision
. IVR/RADIUS AAA compliant (Multilingual IVR)
. SelectNetT provides superior voice quality
. Integrated H.323 gatekeeper and SIP B2BUA for Survivability**

The Quintum® Tenor® AX Series gives businesses with analog voice infrastructures an easy, cost-effective way to capitalize on the power of Voice over IP (VoIP). The Tenor integrates a gateway, a gatekeeper, and intelligent call routing, and supports QoS all in one solution. With its MultiPath architecture, the Tenor connects to the data network through a 10/100 Ethernet interface, the voice network through either a PBX or phone, and the public switched telephone network.

Patented SelectNetT Technology assures high quality voice.
The Tenor employs its patented SelectNetT Technology to monitor calls for jitter, packet loss, and latency, and can transparently switch mid-call to the voice network whenever conditions demand.

More intelligence means greater flexibility.
With its MultiPath Call Routing, the Tenor can intelligently route calls between the PBX, the PSTN, and the IP network to achieve the best combination of cost and quality. The Tenor can also route calls over IP to reduce costs, and then transparently "hop off" to the PSTN, to reach off-net locations. No other VoIP solution can match this flexibility.

More intelligence means easier installation.
With its MultiPath architecture, the Tenor is the only VoIP solution that can be installed without upgrades to the existing voice or data networks. Tenor connects to the data network through a 10/100 Ethernet interface, and to the enterprise and public voice network through a T1, E1, or PRI interface.

More intelligence means greater reliability.
The Tenor is designed to pass calls through to the existing voice network in the event of system malfunction - even a total power failure.

More intelligence means less network congestion.
With its PacketSaverT Technology, the Tenor reduces bandwidth consumption up to 57%, by combining voice packets from several calls into a single packet to minimize bandwidth requirements.

More intelligence means better management
The Tenor comes with a set of easy-to-use configuration and management tools, including Tenor Configuration Manager (GUI) and Wizard, and the Tenor Monitor that offers realtime monitoring of alarms, call status and CDRs. Tenors can easily be remotely managed behind NAT firewalls utilizing the Tenor Remote Management Session Server.*

Quintum VoIP Products

Tenor AF Series

. 2, 4, 6 or 8 Analog line and trunk interfaces
. Up to 8 simultaneous VoIP calls
. Available in MultiPath or Gateway configurations
. Perfect for small enterprises, SOHO and branch office locations
. Support for external Quintum Tenor Monitor
. Support for external Remote Management Session Server*

Intelligent VoIP Access Solutions for Small Enterprises and Remote Locations

Both versions of Tenor still include the features that made Quintum the VoIP market value leader.

. MultiPath architecture for easy integration with existing voice and data infrastructure, meaning little or no re-programming of the PBX, or upgrades are required and no need for special dialing plans.

. Transparent MultiPath Call Routing to intelligently route calls between the PBX, the PSTN, and the IP network to achieve the best combination of cost and quality. The Tenor can also route calls over IP to reduce costs, and then transparently "hop off" to the PSTN, to reach off-net locations.

. SelectNetT Technology to monitor QoS of calls and transparently re-route active calls to the PSTN in real time if voice quality is in jeopardy. The user doesn't even notice the transition.

. PacketSaverT Technology multiplexing to reduce bandwidth consumption by up to 57% by combining voice packets from multiple calls into a single packet.

. Universal Dial Plan that provides a programmable dial plan so Tenor to be integrated into any network environment.

. NATAccessT to allow Tenor to operate behind NAT firewalls to translate internal IP addresses into public addresses when a VoIP call is established with an outside party.

. Remote Management for anywhere, anytime remote management even behind NAT firewalls with Quintum's Remote Management Session Server.

Tenor and Tenor S now assure that VoIP can be deployed easily in existing voice and data networks, offering unmatched voice quality and survivability, network-wide scalability, and easy, secure remote management for installation, configuration, upgrades, troubleshooting and repair. All this adds up to lower TCO than any other VoIP solution on the market!

Quintum VoIP Products

TEnor CMS Series

A more intelligent way to enter the VoIP market

Tenor CMS offers:
. Available in 2 or 8 slot chassis
. Up to 32 T1/E1/PRI Spans per chassis (up to 960 DS0s)
. Up to 960 VoIP channels
. Integrated H.323 gatekeeper
. Intelligent Call Routing
. VoIP and Tandem Circuit Switching
. IVR/Radius AAA Compliant (Multilingual IVR)
. SelectNetT Auto-Switching provides superior voice quality
. Integrated H.323 gateway and gatekeeper or SIP User Agent
. Support for external Quintum Call Routing Server*
. Support for external Quintum Tenor Monitor
. Support for external Remote Management Session Server*

The Quintum Tenor Carrier MultiPath Switch offers a two or an eight port chassis to give service providers, CLECs, ISPs and next-generation service providers an easy, cost-effective way to deliver low-cost, high-quality voice services over IP networks. The Tenor supports H.323 or SIP, intelligent call routing, and QoS all in one solution. Tenor CMS provides the functionality required to support applications such as:

Wholesale VoIP Termination
Call Centers
Least Cost Routing
Tandem Switching
VoIP Local Access
Calling Cards

Carrier class intelligence delivers a complete application solution.

Integrated VoIP and Circuit Switching: Complete VoIP/circuit switch intelligently switches calls between circuits (DS0s) and IP, between multiple circuits and between IP endpoints.
Advanced Call Routing Support: Quintum's external VoIP Call Routing Server provides scalable, centralized network routing control and administration for larger networks. Provides enhanced network-wide routing flexibility including QoS routing, least cost routing, source based routing, and extensive network routing statistics/report generation.
Network Management Support: Quintum's external Network Management Server helps lower maintenance costs through centralized provisioning, alarm monitoring, CDR collecting and real time call monitoring.
Enhanced Interactive Voice Response (IVR): Supports multiple user selectable languages and prompts.

SelectNetT intelligence takes the risk out of VoIP.
With our patented SelectNetT Technology, the Tenor continuously monitors the IP network on a call by call basis for jitter, latency and packet loss, and transparently switches customer calls to the PSTN when required. This intelligent switching allows you to achieve the Quality of Service targets that your customers demand, at costs that traditional phone companies simply cannot match.

PacketSaverT intelligence minimizes network congestion.
PacketSaverT Technology allows the Tenor to reduce bandwidth consumption up to 50%, by combining voice packets from several calls into a single packet to minimize bandwidth overhead.

NATAccessT intelligence means greater security.
The Tenor VoIP MultiPath Switch also features a unique technology that allows it to operate behind NAT-enabled firewalls. The innovative NATAccessT solution overcomes the problem of NAT firewalls not correctly translating internal IP addresses into public addresses when a VoIP call is established with an outside party.

Support for SS7/C7.
Tenor CMS is able to support SS7/C7 using an external signaling gateway*.

Quintum VoIP Products

Tenor Call Relay Session Border Controller

A more intelligent way to enter the VoIP market

Tenor Call Relay offers:
. Enterprise and High Capacity Service Provider (SP) versions
. Supports up to 672 simultaneous VoIP calls
. Supports H.323 and SIP Protocols
. Provides H.323/SIP Signalling Translation and Inter-networking
. Local H.323 Gatekeeper Options
. Provides VoIP switching between IP networks
. Supports IP endpoints behind the NAT firewall and remote NAT
. PacketSaverT reduces bandwidth requirements 57%*

Quintum's Tenor Call RelayT Session Border Controllers provide a VoIP conduit between IP networks allowing for end to end VoIP communications across multiple IP networks. All calls are switched through multiple IP networks with just one single compression and decompression of the voice. The result is less latency and higher voice quality as the call passes from network to network.

The Tenor Call Relay allows VoIP endpoints, such as VoIP gateways, IP phones and IP soft phones which are behind a Network Address Translation (NAT) firewall, to communicate with VoIP networks on other external IP networks. This allows both enterprises and service providers to expand their VoIP networks to home offices, branch offices, customers, partners, and across the public Internet.

Tenor Call Relay also provides a single point for call management, administration and security at the edge of your VoIP network.

With this unique intelligent VoIP network switching, Tenor Call Relay makes expanding VoIP calling both easy and risk free. Call Relay is available in two versions: Enterprise Call Relay 60 supports up to 60 simultaneous VoIP calls; Call Relay SP supports up to 672.

More intelligence means higher voice quality and less delay
When separate VoIP networks are linked using circuit switched connections, the multiple compression and de-compression processes that occur increase transmission delay and cause unnecessary degradation of the voice quality. Tenor Call Relay eliminates these delays with its direct IP connection between VoIP networks, thereby increasing voice quality.

More intelligence means greater efficiency and security
Tenor Call Relay comes with a Gatekeeper option* for registration of IP endpoints and phones within the local zone. Calls within the zone are routed directly between endpoints; Calls out of the zone pass through the Call Relay for maximum security.

More intelligence means less network congestion
With its PacketSaverT* Technology, the Tenor Call Relay reduces bandwidth consumption of VoIP calls by multiplexing multiple calls into the same packet, reducing the overall bandwidth utilization by up to 57%, beyond voice compression and silence suppression.

Tenor MultiPath Switch vs. Traditional VoIP Gateways

Benefits	Tenor MultiPath	Traditional VoIP Gateway
Easy to Deploy	? MULTIPATH ARCHITECTURE provides easy integration with customer's existing network architecture. ? Utilizes trunks with STANDARD TELEPHONY INTERFACES to integrate "legacy devices" when adding VoIP services.	? HIDDEN COSTS for additional tie trunk hardware on the PBX. ? PBX MUST be REPROGRAMMED to route VoIP calls. ? ADDS COMPLEXITY and COST to deployment.
Intelligent Call Routing and Switching	? Tenor provides DYNAMIC CALL ROUTING and SWITCHING intelligence, thus enabling Tenor to be deployed TRANSPARENTLY to the existing network. ? Supports INTEGRATED CIRCUIT SWITCHING and VoIP CALLS in the same device.	? NO CALL ROUTING or switching. Calls are are simply converted from circuit to packet and back again. ? NO CIRCUIT SWITCHING - all calls must be switched externally.
Survivable and Reliable	? FAILOVER TO PSTN if IP network experiences QoS or connectivity issues. ? BTBUA (SIP) offers "LOCAL PROXY" for support of SIP endpoints/phones in branch/remote/small offices.	? NO CALL ROUTING or switching. Calls are are simply converted from circuit to packet and back again. ? NO CIRCUIT SWITCHING - all calls must be switched externally.
Quality of Service (QoS)	? SelectNetT provides SEAMLESS MID-CALL TRANSITION TO THE PSTN if the IP network experiences an issue that affects call quality. ? PacketSaverT REDUCES CALL BANDWIDTH , minimizing network congestion.	? END-TO-END QoS IS NOT AVAILABLE unless a complete QoS is implemented throughout the network. ? If a fallback option exists, the calls may ONLY ROUTED TO THE PSTN AT CALL SETUP .
Flexible	? HIGHLY PROGRAMMABLE DIAL PLAN allows simplified integration with customer's internal/external dial plan requirements. ? INTEGRATED CALL ROUTING & PROCESSING allows the Tenor to support a complete VoIP network independently.	? TRADITIONAL GATEWAYS LACK INTELLIGENCE and simply provides circuit-to-packet conversion. ? NO ABILITY TO SUPPORT CALL ROUTING , thus requiring external intelligence to support a VoIP application, which increases cost and complexity.
Secure	? NATAccessT technology allows the Tenor to securely operate behind and with the customer's firewall.	? Gateways typically need to sit OUTSIDE THE FIREWALL in the DMZ where it's far LESS SECURE .

VoIP Products

IP Phones
IP Gateways
- Adtran
- Grandstream
- Digium
- Quintum
IP Routers and CPE