VOIP Gateways
Showing 1–16 of 23 results
-
Grandstream HT812 FXS ATA, 2 Port Voip Gateway, Dual GbE Network
$85.80The HT812 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analogue devices to a manageable and powerful VoIP network. Built using Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments. The HT812 comes with 2 easy-to-use FXS ports, an integrated Gigabit NAT router, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance.
• Supports 2 SIP profiles through 2 FXS ports and dual Gigabit ports
• Includes a built-in NAT router which can handle routing speeds up to 100MBps
• TLS and SRTP security encryption technology to protect calls and accounts
• Automated provisioning options include TR-069 and XML config files
• Supports 3-way voice conferencing
• Failover SIP server automatically switches to secondary server if main server loses connection
• Supports T.38 Fax for creating Fax-over-IP
• Supports a wide range of caller ID formats
• Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning -
Grandstream HT801 1 Port FXS ATA
$99.00The HT801 is a single port analog telephone adapter (ATA) that allows users to create a high-quality and manageable IP telephony solution for residential and office environments.
The HT801 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable and powerful VoIP network. Built upon Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments. The HT801 comes with 1 easy-to-use FXS ports, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance.
• Supports 1 SIP profile through a single FXS port and a single 10/100Mbps port
• TLS and SRTP security encryption technology to protect calls and accounts
• Automated provisioning options include TR-069 and XML config files
• Supports 3-way voice conferencing
• Failover SIP server automatically switches to secondary server if main server loses connection
• Supports T.38 Fax for creating Fax-over-IP
• Supports a wide range of caller ID formats
• Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning
• Supports advanced telephony features, including call transfer, call forward, call-waiting, do not disturb, message waiting indication, multilanguage prompts, flexible dial plan and more -
Grandstream HT802 2 Port FXS ATA
$99.00The HT802 is a 2 port analog telephone adapter (ATA) that allows users to create a high-quality and manageable IP telephony solution for residential and office environments.
The HT802 is a 2 port analog telephone adapter (ATA) that allows users to create a high-quality and manageable IP telephony solution for residential and office envir onments. Its ultra-compact size, voice quality, advanced VoIP functionality, security protection and auto provisioning options enable users to take advantage of VoIP on analog phones and enables service provider s to offer high quality IP service. The HT802 is an ideal ATA for individual use and for large scale co mmercial IP voice deployments.
• Supports 2 SIP profiles through 2 FXS ports and a single 10/100Mbps port
• TLS and SRTP security encryption technology to protect calls and accounts
• Automated provisioning options include TR-069 and XML config files
• Supports 3-way voice conferencing
• Failover SIP server automatically switches to secondary server if main server loses connection
• Supports T.38 Fax for creating Fax-over-IP
• Supports a wide range of caller ID formats
• Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning -
Cisco SPA112 2 Port Phone Adapter 2X RJ11 FXS, 1X LAN 10/100
$129.00Product Overview
The Cisco SPA112 2 Port Adapter enables high-quality VoIP service with a comprehensive feature set through a broadband Internet connection. Easy to install and use, it works over an IP network to connect analog phones and fax machines to a VoIP service provider and provides support for additional LAN connections.
The Cisco SPA112 includes two standard telephone ports to connect existing analog phones or fax machines to a VoIP service provider. Each phone line can be configured independently. With the Cisco SPA112, users can protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines as well as control their migration to IP voice with an extremely affordable, reliable solution.
Compact in design and compatible with international voice and data standards, the Cisco SPA112 can be used with residential, home-office, and small business VoIP service offerings, including full-featured hosted or open source IP PBX environments. This easy-to-use solution delivers advanced features to better connect employees and serve customers, all on a highly secure Cisco network.
The Cisco SPA112 2 Port Adapter:
• Enables high-quality VoIP service with a comprehensive feature set through a broadband Internet connection
• Provides high-quality, clear-sounding voice, using advanced voice quality-of-service (QoS) capabilities and the industry-leading voice Session Initiation Protocol (SIP) stack
• Supports reliable faxing with simultaneous voice and data use
• Includes two standard telephone ports, each with an independent phone number, for use with fax machines or analog phone devices
• Is compatible with all industry voice and data standards and common telephone features such as caller ID, call waiting, and voicemail
• Includes a simple-to-use web-based configuration utility for easy deployment -
Cisco SPA122 ATA with Router
$129.00Product Overview
The Cisco SPA122 ATA with Router combines VoIP services with an internal router for LAN connectivity. Easy to install and use, it works over an IP network to connect analog phones and fax machines to a VoIP service provider and provides support for additional LAN connections.
The Cisco SPA122 includes two standard telephone ports to connect existing analog phones or fax machines to a VoIP service provider. It also includes two 100BASE-T RJ-45 Ethernet ports for WAN and LAN connectivity. Each phone line can be configured independently. With the Cisco SPA122, users can protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines, as well as control their migration to IP voice with an extremely affordable, reliable solution.
Compact in design and compatible with international voice and data standards, the Cisco SPA122 can be used with residential, home-office, and small business VoIP service offerings, including full-featured hosted or open source IP PBX environments. This easy-to-use solution delivers advanced features to better connect employees and serve customers, all on a highly secure Cisco network.
The Cisco SPA122 ATA with Router (Figures 1 and 2):
● Enables high-quality VoIP service with a comprehensive feature set through a broadband Internet connection
● Provides high-quality, clear-sounding voice, using advanced voice quality-of-service (QoS) capabilities and the industry-leading voice Session Initiation Protocol (SIP) stack
● Supports reliable faxing with simultaneous voice and data use
● Includes two standard telephone ports, each with an independent phone number, for use with fax machines or analog phone devices, and one fast Ethernet WAN port, and one fast Ethernet LAN port for local home or business network connection
● Is compatible with all industry voice and data standards and common telephone features such as caller ID, call waiting, and voicemai -
Grandstream HT814 FXS ATA, 4 Port Voip Gateway, Dual GbE Network
$165.00The HT814 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analogue devices to a manageable, robust network. Built using Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments. The HT814 comes with 4 easy-to-use FXS ports, an integrated Gigabit NAT router, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance.
• Supports 2 SIP profiles through 4 FXS ports and dual Gigabit ports
• Includes a built-in NAT router which can handle routing speeds up to 100MBps
• TLS and SRTP security encryption technology to protect calls and accounts
• Automated provisioning options include TR-069 and XML config files
• Supports 3-way voice conferencing
• Failover SIP server automatically switches to secondary server if main server loses connection
• Supports T.38 Fax for creating Fax-over-IP
• Supports a wide range of caller ID formats
• Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning -
Grandstream HT818 FXS ATA, 8 Port Voip Gateway, Dual GbE Network
$249.48The HT818 is a powerful 8-port VoIP gateway with 8 FXS ports and an integrated Gigabit NAT router. Built for users looking for a strong analogue-to-VoIP converter, it features Grandstream’s market-leading SIP ATA/gateway technology with millions of units successfully deployed worldwide. This powerful gateway carries exceptional voice quality in various application environments, strong encryption with unique security certificate per unit, automated provisioning for volume deployment and device management, and outstanding network performance for enterprise use.
• Supports 2 SIP profiles and 8 FXS ports
• Strong AES encryption with security certificate per unit
• Automated & secure provisioning options using TR069
• 3-way voice conferencing per port
• Exceptional voice quality with wide-band HD codec
• Supports T.38 Fax for reliable Fax-over-IP
• Supports dual Gigabit network ports
• High performance NAT router -
Grandstream GXW4104 4 FXO Port VoIP gateway with dual 10/100
$399.00Description
The GXW410x FXO gateway series enables businesses of all sizes to create and deploy a VoIP and analog hybrid solution. Make deployments easy by seamlessly connect multiple locations and all devices within an office to any hosted or on premise IP PBX network.Features
4 or 8 ports
2 10/100 Mbps network ports
Comprehensive codec support, caller ID, flexible dial plans and security protection
Advanced security protection with SRTP
Designed and tested for full interoperability with leading IP-PBXs, soft-switches and SIP-based environments
Failover SIP server feature in case main SIP server goes down -
Beronet 2x BRI/S0 2x FXS Modul 2x BFTAdapters, 1x BFBridge
$438.90Overview
The BF2S02FXS Hybrid Module is a 2 Port BRI and 2 Port FXS Module for the beroNet VoIP Gateways
Each BRI port of the BF2S02FXS can be configured as NT (Network Termination) or TE (Terminal Equipment). In both modes, each BRI port can operate in Point-to-Multi-Point (PMP) or in Point-to Point (PTP) mode. The switching of the TE / NT mode and the switch off the termination resistors can easily be set in the software (Jumper Free) which makes the troublesome reposition jumpers unnecessary. Additionally to the above mentioned 2 BRI Ports the BF2S02FXS provides 2 ports FXS (Foreign Exchange Station) for the beroNet VoIP Gateways. FXS interfaces are used to connect devices like analog phones or faxmachines to your beroNet.
Specifications
· 2 BRI (S0) ISDN ports
· 2 analog FXS ports
· each BRI Port configurable as NT or TE with PIN interchanging
· termination resistors (100 ohm) per line
· TE/NT mode and termination is selectable via software (Jumperfree)
· layer 2 is Q.921 and layer 3 is Q.931 (EuroISDN DSS1) compatible
· DSS1 feature set: CLIP/No-Screening, CLIR, COLP, UUS, MCID, CD, CNIP -
Patton SN4112/JO SmartNode Dual FXO VoIP Gateway 1×10/100baseT, H.323 and SIP, External Power
$495.00• Up to 8 FXS and/or FXO ports
– Compact, reliable stand-alone VoIP gateway with different port options. Supports simultaneous voice or fax calls on all ports.
• Advanced Local Call Switching
– Virtual interfaces and routing tables provide industry leading flexibility in call handling programming. Local call switching, soft fallback to alternative routes. Simultaneously connects to multiple SIP services/IP PBXs.
• Complete SIP and T.38 support
– Supports the complete range of industry standard VoIP: SIP, H.323, T.38 fax, fax and modem handling, DTMF relay. Codecs G.729, G.723, etc.
• Easy Management & Provisioning
– Web-based management, SNMP, command line interface. Automated mass provisioning for efficient large-scale deployments.
• Outstanding Interoperability
– Proven integration for voice and T.38 fax with 3CX®, Asterisk™, PingTel™ and other leading IP PBX systems and soft switch vendors.
• Supported by SmartNode™ Redirection Service -
Grandstream GXW4108 – 8 FXO Port VoIP gateway with dual 10/100
$500.28Description
The GXW410x FXO gateway series enables businesses of all sizes to create and deploy a VoIP and analog hybrid solution. Make deployments easy by seamlessly connect multiple locations and all devices within an office to any hosted or on premise IP PBX network.Features
4 or 8 ports
2 10/100 Mbps network ports
Comprehensive codec support, caller ID, flexible dial plans and security protection
Advanced security protection with SRTP
Designed and tested for full interoperability with leading IP-PBXs, soft-switches and SIP-based environments
Failover SIP server feature in case main SIP server goes down -
Patton SN4112/JS SmartNode Dual FXS VoIP Gateway 1×10/100baseT, H.323 and SIP, External Power
$599.01• Up to 8 FXS and/or FXO ports
– Compact, reliable stand-alone VoIP gateway with different port options. Supports simultaneous voice or fax calls on all ports.
• Advanced Local Call Switching
– Virtual interfaces and routing tables provide industry leading flexibility in call handling programming. Local call switching, soft fallback to alternative routes. Simultaneously connects to multiple SIP services/IP PBXs.
• Complete SIP and T.38 support
– Supports the complete range of industry standard VoIP: SIP, H.323, T.38 fax, fax and modem handling, DTMF relay. Codecs G.729, G.723, etc.
• Easy Management & Provisioning
– Web-based management, SNMP, command line interface. Automated mass provisioning for efficient large-scale deployments.
• Outstanding Interoperability
– Proven integration for voice and T.38 fax with 3CX®, Asterisk™, PingTel™ and other leading IP PBX systems and soft switch vendors.
• Supported by SmartNode™ Redirection Service
– A free service enabling zero-touch mass deployments for Service Providers and Distributors with auto-provisioning serversCapacity
• Up to 8 simultaneous VoIP or T.38 fax calls (depending on the model)
Voice Signaling
• H.323v4, SIPv2 (B2BUA capable, multi-instance, simultaneous support of multiple registrars and direct IP dialing)
• SIP call transfer, redirect
• DTMF in-band & out-of-band
• All tones programmable (dial, ringing, busy)
Voice Processing
• CODEC G.711 a-law/mu-law, G.723, G.729ab
• G.726, G.727. T.38 fax relay (9.6 k, 14.4 k)
• G.711 transparent fax and bypass
Call Switching and Services
• Virtual interfaces
• Regular expression based call routing and number manipulation
• Number blocking
• Short-dialing
• Digit collection, distribution and hunt groups
• Transparent line extension
• Fallback Routing: Soft fallback to alternative route(s)
FXS Connectivity
• 2-wire Loopstart on RJ-11/12
• Short haul loop 1.1km @3REN
• EuroPOTS (ETSI EG201188)
• Programmable AC impedanc -
Patton SN4114/JO SmartNode 4 FXO VoIP Gateway 1×10/100baseT, H.323 and SIP, External Power
$619.08• Up to 8 FXS and/or FXO ports
– Compact, reliable stand-alone VoIP gateway with different port options. Supports simultaneous voice or fax calls on all ports.
• Advanced Local Call Switching
– Virtual interfaces and routing tables provide industry leading flexibility in call handling programming. Local call switching, soft fallback to alternative routes. Simultaneously connects to multiple SIP services/IP PBXs.
• Complete SIP and T.38 support
– Supports the complete range of industry standard VoIP: SIP, H.323, T.38 fax, fax and modem handling, DTMF relay. Codecs G.729, G.723, etc.
• Easy Management & Provisioning
– Web-based management, SNMP, command line interface. Automated mass provisioning for efficient large-scale deployments.
• Outstanding Interoperability
– Proven integration for voice and T.38 fax with 3CX®, Asterisk™, PingTel™ and other leading IP PBX systems and soft switch vendors.
• Supported by SmartNode™ Redirection Service
– A free service enabling zero-touch mass deployments for Service Providers and Distributors with auto-provisioning serversCapacity
• Up to 8 simultaneous VoIP or T.38 fax calls (depending on the model)
Voice Signaling
• H.323v4, SIPv2 (B2BUA capable, multi-instance, simultaneous support of multiple registrars and direct IP dialing)
• SIP call transfer, redirect
• DTMF in-band & out-of-band
• All tones programmable (dial, ringing, busy)
Voice Processing
• CODEC G.711 a-law/mu-law, G.723, G.729ab
• G.726, G.727. T.38 fax relay (9.6 k, 14.4 k)
• G.711 transparent fax and bypass
Call Switching and Services
• Virtual interfaces
• Regular expression based call routing and number manipulation
• Number blocking
• Short-dialing
• Digit collection, distribution and hunt groups
• Transparent line extension
• Fallback Routing: Soft fallback to alternative route(s)
FXS Connectivity
• 2-wire Loopstart on RJ-11/12
• Short haul loop 1.1km @3REN
• EuroPOTS (ETSI EG201188)
• Programmable AC impedanc -
Beronet PCIe 16-64 Ch Basebd Supports 16-64 Concurrent Chan
$713.90Overview
The beroNet PCIe cards are full-fledged VoIP Gateway Media Gateways in the form factors PCI Express. They are modular and can be equipped with up to two modules. Digital ISDN (BRI / PRI), analog (FXS / FXO) and GSM modules are available for the beroNet Gateways. An optional PCM-Bus connection between two Gateways or beroNet card also ensures you real hardware bridging for transparent audio transmission of sensitive voice, video, data and Fax services. The beroNet PCIe cards are ideal for PBX manufacturers, Telefony Appliances, Fax appliances, Unified Communication Appliances, and hybrid systems.
Specifications
· Codecs: G.723.1 and Annex A, G.729 a/b, G.726 (up to 32 channels),
· G.711 u/a (up to 128 channels)
· G.168/G.165 echo cancellation with echo path change detection, up to 128ms
· Voice activity detection / comfort noise generation
· DTMF digit detection and generation
· T.38 Fax Relais (V.27, V.29 and V.17) (up to 16 channels)
· SIP User Agent IETF RFC3261 conform
· -
Beronet PCIe PRI/E1 Module For Beronet Baseboard
$823.90Overview
The BF1E1 Module can be configured individually to NT (Network Termination) or TE (Terminal equipment) mode. The default Pin-Out of this module is always TE mode with used PINS 1,2,4,5. If you want to use NT-mode you may need a “cross cable” which is optional available (BFE1Cross). Line termination (120 ohms) is selectable by DIP switches on the module.Specifications
· 1 PRI (S2M) port
· configurable as NT or TE
· termination resistors 120 Ohm switchable via DIPs
· layer 2 is Q.921 and layer 3 is Q.931 (EuroISDN DSS1) compatible
· DSS1 feature set: CLIP/No-Screening, CLIR, COLP, UUS, MCID, CD, CNIP
· Q.SIG feature set: CNIP
· CE and TBR-4 compliance (ISDN PRI)
Dimensions and Weight
- 1
- 2