Hardware

Analog VoIP gateways

Linksys SPA 2102

- Dual ports to connect existing analog phones or fax machines, or to a PBX system
- Two 100BaseT RJ45 Ethernet interfaces (LAN-WAN) to connect to a home or office LAN, as well as an - Ethernet connection for a broadband modem or router (WAN)
- 2 RJ-11 FXS phone ports-for analog circuit telephone device
- Independent configuration of each line using software controlled by the service provider or the end user
- Strong security with reliable, encryption-based methods for communication, provisioning, and servicing.
- Protocol support: SIP


Linksys SPA3102

- One RJ-11 POTS (Plain Old Telephone Service) FXS port to connect an existing analog phone or fax machine

- One PSTN FXO port to connect to a Telco or PBX circuit

- Two 100BaseT RJ-45 Ethernet interfaces to connect to a home or office LAN, and an Ethernet connection to a broadband modem or router

- FXS and FXO lines that can be independently configured

- Secure encryption-based methods for communication, provisioning, and servicing

- Protocol support: SIP

Patton 4022

- Dual RJ-11 FXS ports to connect existing analog phones or fax machines, or to a PBX system
- Two 100BaseT RJ45 Ethernet interfaces (LAN-WAN)
- Protocol support: SIP


Grandstream HT 704

- One 100BaseT RJ45 Ethernet interfaces

- 4 RJ-11 FXS ports to connect existing analog phones or fax machines to a PBX system

- Protocol support: SIP

Grandstream GXW4008

- 8 RJ-11 FXS ports to connect existing analog phones or fax machines to a PBX system
- Two 100BaseT RJ45 Ethernet interfaces (LAN-WAN)
- Protocol support: SIP


Linksys SPA8000

- 8 RJ-11 FXS ports to connect analog telephones to IP-based data networks
- A single multiport RJ-21 50-pin connector, offering an alternative connection choice
- One 10/100 Base-T RJ-45 Ethernet interface to connect to either a router or multilayer switch
- Toll-quality voice and carrier-grade feature support
- Large-scale deployment management
- Strong security with reliable, encryption-based methods for communication, provisioning, and servicing
- Protocol support: SIP


Linksys SPA8800

- 4 RJ-11 FXS ports to connect analog telephones to IP-based data networks

- 4 RJ-11 FXO ports to connect PSTN lines

- One 10/100 Base-T RJ-45 Ethernet interface

- Converts voice traffic into data packets for transmission over an IP network

- Session Initiation Protocol (SIP) standards for voice and data networking provides reliable voice and fax operation

- Secure, encryption-based methods for communicating, provisioning, and servicing

- Toll-quality voice and carrier-grade feature support

- Large-scale deployment management

Digital VoIP gateways

Audiocodes Mediant 600

- 4/8 RJ-45 BRI ports per gateway

- 4 FXS ports using RJ-11 conn.

- Dual 10/100 Base-T RJ-45 Ethernet interface

- RS-232 for configuration and troubleshooting

- Protocol support: SIP

Audiocodes Median 1000

- 4 BRI ports (8 calls) per module, up to 5 modules per gateway with S/T interfaces.

Supports Euro ISDN, NI2, 5ESS or QSIG.

- 4 FXO/FXS ports using RJ-11 connectors per module; Up to 6 modules per gateway, Ground Start and Loop Start.

- 1, 2 or 4 E1/T1/J1 spans using RJ-48c connectors per module. Up to 4 digital modules (maximum 4 spans per gateway). Optional 1+1 or 2+2 fallback spans.

- Voice interface: Equipped with 6 Slots that can host voice modules.

Up to a maximum of 24 analog ports or 4 digital spans.

Digium G100

- One T1/E1 port

- One 10/100 Base-T RJ-45 Ethernet interface
- 30 VoIP channels






VoIP Phones

Digium D40

- 2 SIP lines
- Dual 10/100 Ethernet
- 3.5' LCD
- PoE support
- HD voice
- Full duplex speaker phone


Digium D70

- 6 SIP lines
- Dual 10/100 Ethernet
- 4.5' LCD
- PoE support
- HD voice
- Full duplex speaker phone
- 10 Feature Keys
- 10 Rapid dial keys



Cisco SPA512g

- Pixel-based display: 128 x 64 monochrome LCD graphical display with backlight

- Dedicated illuminated buttons for:

--- Audio mute on/off

--- Headset on/off

--- Speakerphone on/off

- 4 way rocking directional knob for menu navigation

- Voicemail message waiting indicator light

- Voicemail message retrieval button

- Dedicated hold button

- Settings button for access to feature, setup, and configuration menus

- Volume control rocking up/down knob controls handset, headset, speaker, ringer

- Standard 12-button dialing pad

- High-quality handset and cradle

- Built-in high-quality microphone and speaker

- Headset jack: 2.5 mm

- Electronic Hook Switch (EHS) Support with selected Plantronics headsets with adapter

- Gigabit Switch Port and PC Port: 1000BASE-T RJ-45

- 802.3af compliant Power over Ethernet (PoE)

Grandstream GXP1450

- 180x60 pixel backlit graphical LCD display with up to 4 level grayscale

- 2 dual-color line keys (with 2 SIP accounts and up to 2 all appearances) 

- 3 XML programmable context-sensitive soft keys, 3-way conference

- HD wideband audio, superb full-duplex hands-free speakerphone with advanced acoustic echo cancellation and excellent double-talk performance

- Large phonebook (up to 500 contacts) and call history (up to 500 records)

- Automated personal information service (e.g., local weather, etc), personalized music ring tone/ring back tone, flexible customizable screen content & format using XML, and advanced Web and enterprise applications integration (pending)

- Dual switched auto-sensing 10/100Mbps network ports with integrated PoE

- Automated provisioning using TR-069 or encrypted XML configuration file, SRTP{ and TLS for advanced security protection, 802.1x for media access control

Grandstream DP715

- DECT base station registers up to 5 DECT handsets and talks to up to 4 handsets concurrently

- When multiple handsets share the same SIP account, Hunting Group supports the following flexible options:

Linear Mode, all phones ring sequentially in the predestinated order 

Parallel Mode, all phones ring concurrently and after one phone answers,the remaining available phones can place new calls 

Shared Line Mode, all phones ring concurrently and always share the same line similar to analog phones

- Advanced telephony features including Caller ID, Call Waiting, 3-Way Conference, Transfer, Forward, Do Not Disturb, Message Waiting Indication, auto answer, multi-language voice prompt, flexible dial plan

- Support comprehensive voice codecs including G.711, G.723.1, G.729A/B, G.726 and iLBC

- Secure and automated provisioning using HTTP/HTTPS/Telnet/TFTP, multiple SIP accounts, SIP over TCP/TLS, SRTP

- Multi-Languages - English, German, French, Spanish, Dutch, Italian, Czech, Danish, Greek, Norwegian, Polish, Portuguese, Russian, Swedish, Turkish

- Currently Pending - TR069, IPV6(pending)

Snom 720

- 4-line B/W display

- 18 LED function keys

- 12 identities

- Wideband audio

- Hands-free operation

- Power over Ethernet (PoE)

- WLAN / Bluetooth Headset ready

- VLAN

- 2 x Gigabit LAN ports

Yealink T38G

- TI Aries chipset and TI voice engine 

- Dual-port Gigabit Ethernet (Router & Switch) 

- Supports IPV6 

- Power over Ethernet 

- 4.3” TFT-LCD, 480 x 272 pixel, 16.7M colors 

- 6 VoIP accounts, Hotline, Emergency call

- Soft keys programmable

- Supports up to 6 expansion modules(EXP39)

- Supports Wireless Headset Adapter(EHS36)

- 2xRJ45 10/100/1000Mbps Ethernet ports

- 4.3” TFT-LCD, 480 x 272 pixel, 16.7M colors

- 48 keys including 16 programmable keys