VoIP Accessories

The Grandstream GXP1610 IP Phone is a simple-to-use IP phone for small-to-medium businesses (SMBs) and home offices. This Linux-based model features a 1 (single) SIP account, up to 2 call appearances and 3 XML programmable soft keys. A 132x48 LCD screen creates a clear display for easy viewing. Additional features such as dual switched 10/100 Mbps ports, multi-language support, Electronic Hook Switch support for Plantronics headsets, call-waiting and 3-way conferencing allow the GXP1610 to be a high quality, user-friendly and dependable IP phone. The GXP1610 is No PoE Integrated.

"New Series"

The Grandstream DP730 is a DeCT Handset IP Phone can be used with the DP750, DP752 & DP755 DECT Base Station.

Utilize mobility and call efficiency to increase the productivity throughout any communications network. The DP730 mobile DECT handset is the perfect solution for any business, warehouse, retail store or residential environment. It is supported by Grandstream’s DP750, DP752,  and DP755 DECT VoIP base station.

• Supports a range of up to 400 meters outdoors and 50 meters indoors when used with the DP752 & DP755 (300 meters outdoors when used with DP750)
• 2.4 inch (240x320) color LCD with 3 programmable soft keys
• Offers 40-hour talk time and 500-hour standby time
• Supports up to 10 SIP accounts and 10 lines per handset as well as 3-way conferencing
• Dedicated push-to-talk button, built-in proximity and accelerometer sensors for gesture control and alert notification
• HD audio on the speakerphone, handset, and headset jack
• Seamless 1-touch door control with Grandstream’s GDS series of Facility Access devices
• Software and firmware updates over-the-air

The GSC3505 is a 1-way public address SIP speaker that allows offices, schools, hospitals, apartments and more to build powerful public address announcement solutions that expand security and communication.

• Speaker with HD acoustic chamber
• Features SIP paging, multicast paging, callwaiting with priority override
• Full-band and wide-band audio codec support including Opus & G.722
• Integrated dual-band 802.11 a/b/g/n WiFi and Bluetooth
• One 10/100Mbps network port with PoE/ PoE+

• HD audio options
• Up to 1.5m of audio input
• Up to 3m of audio output
• Built-in RFID chip reader for keyless entry
• Supports SIP calls to IP phones to facilitate communication
• Built with a metal casing to make it weatherproof and vandal resistant
• Integrated PoE to power the device and provide a network connection
• Built-in dual microphone and HD loudspeaker with advanced AEC offers intercom functionality

Grandstream UCM6304 Key Features

• Built in 4 FXO ports (PSTN Line) and 4 FXS ports (Analog Device); All ports have lifeline capability in case of power outage.
• Up to 15 Simultaneous Video Conference Rooms, up to 200 simultaneous participants in all room combined, up to 9 video feeds in all conference room
• Voice Conference : Up to 200 parties (G.711)
• Supports up to 2000 users and up to 300 concurrent calls
• Up to 200 SIP trunk accounts
• Zero configuration provisioning of Grandstream SIP endpoints
• Built-in conferencing & meetings platform; supports desktop, Wave app, and SIP endpoints
• Wave for Android, iOS, Chrome and Firefox browsers allows communication with all UCM6300 users & solutions
• API available for third-party integrations, including CRM and PMS platforms
• Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
• Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
• Automated NAT firewall traversal service facilitates secure remote connections
• Supports Full-Band Opus voice codec and H.264/H.263/ H.263+/VP8 video codec, jitter resilience up to 50% packet loss
• Compatible with GDMS for cloud setup, management and monitoring
• Based on Asterisk* version 16 open source telephony operating system

Grandstream UCM6308 Key Features

• Built-in 8 FXO ports (PSTN Line) and 8 FXS ports (Analog Device); All ports have lifeline capability in case of power outage.
• Up to 25 Simultaneous Video Conference Rooms, up to 300 simultaneous participants in all room combined, up to 9 video feeds in all conference room
• Voice Conference : Up to 300 parties (G.711)
• Supports up to 3000 users and up to 450 concurrent calls
• Up to 200 SIP trunk accounts
• Zero configuration provisioning of Grandstream SIP endpoints
• Built-in conferencing & meetings platform; supports desktop, Wave app, and SIP endpoints
• Wave for Android, iOS, Chrome and Firefox browsers allows communication with all UCM6300 users & solutions
• API available for third-party integrations, including CRM and PMS platforms
• Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
• Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
• Automated NAT firewall traversal service facilitates secure remote connections
• Supports Full-Band Opus voice codec and H.264/H.263/ H.263+/VP8 video codec, jitter resilience up to 50% packet loss
• Compatible with GDMS for cloud setup, management and monitoring
• Based on Asterisk* version 16 open source telephony operating system