1

Topics:
1. Introduction and history about
Grandstream

2. Definition of Terms

3. UCM Series and its Call Features

4. IP Phones (GRP Series & Video IP

Phones and Wireless IP Phones

5. Example of deployment scenarios

6. Questions and answer

2
WHO IS
GRANDSTREAM
• Company founded in 2002 in Boston, MA.

• +600 employees in 14 countries

(+50% R&D Engineers)
Boston Hangzhou

• More than 100 direct distributors and LA

Dallas
Marruecos

thousands of authorized resellers around the Shenzhen

world. Venezuela

• Global presence with offices in the U.S. (Boston,

Los Angeles, Dallas), Venezuela, Morocco, China
(Shenzen and Hangzhou) and Malaysia

• 4 Research and Development Centers

(Dallas, Los Angeles, Shenzen, and Hangzhou)
Award-Winning History
• Frost & Sullivan’s 2016 Enterprise IP Endpoints Company of the Year

• Named to Deloitte Technology Fast 500 multiple times

• 17 time winner of IT Magazine’s Product of the Year Award

• 11 time winner of TMC Lab’s Innovation Award

• Products trusted by some of the largest service providers in the world

Market Analysis
Grandstream: Worldwide and Regionally

• Number one manufacturer of Low and Medium Density

VoIP ATAs and Analog Gateways

• Each year, Grandstream distributes more than 2 million

units

• Leading IP PBX manufacturer in the small and medium

business market

• Grandstream is one of the top two manufacturers of SIP

video phones

• Grandstream is one of the top three SIP phone

manufacturers
Definition of Terms
FXO port (Foreign Exchange Office)
PSTN / Analog Trunk

FXS port (Foreign Exchange Subscriber)

Analog Endpoints (Analog Phones, Fax Machine)

Note:
Grandstream UCM Series is a Fixed Device / Not Modular.
The capacity and ports cannot be upgraded.

6
Introduction to UCM6xxx Series
The Grandstream UCM IP PBX series deliver an
impressively powerful and easy to manage platform that
brings together different technologies, including voice,
video communications, video surveillance, data tools,
mobility options, and facility access management.
Enterprise-grade features in an affordable, compact, quiet
& easy-to-manage PBX designed specifically for the SMB
and Enterprise markets.

01 No licensing 03 Fast and easy

Fees setup and
management

02 04 Lifetime Firmware
All Hardware and
Software updates included
included
UCM6xxx Components
Voice
- Secure, high-quality and reliable.
- Audio Conference
- Audio Codecs Support
- Voice Features Customization

Video
- SIP Video Calls / Conference.
- Surveillance Integration
- Video Codecs Support
- WebRTC
- IPVideoTalk Integration

Mobility
- Softphone Apps
- Remote monitoring
- Multiple offices connections
- GDMS

Data
- Call Recording
- API Interface
- System Backup / Restore
- Voicemail and IM
- Fax Service
- LDAP / LDAPS Server
- CRM and PMS Integration
- Call Center features and statistics
UCM6510 – IP PBX
IP PBX Appliance

• E1/T1/J1 Interface

• 2 PSTN trunk FXO ports, 2 FXS ports with lifeline

capability

• Up to 2000 SIP endpoints and up to 50 SIP trunk

accounts

• Dual Gigabit network ports, integrated PoE, USB

and SD ports

• Integrated NAT router

• Zero Config endpoint provisioning and no licensing

fees
HA100
High-Availability Controller for UCM6510

• Connects and constantly monitors 2

UCM6510s for high availability

• 14 LED indicators showing real-time status of

telecom lines, network lines, auxiliary devices
etc.

• 10-50 Second System Switching

• Connect the HA100 to PSTN, FXS ports and

UCM6510
UCM6300
SERIES
Unified Communications
& Collaboration Solution
The UCM6300 series provides a high-end unified
communications solution packed with an ecosystem of
mobility, security, video and collaboration tools
UCM6300 Specifications

UCM6301 UCM6302 UCM6304 UCM6308  Based on Asterisk* Version 16.

 1/2/4/8 FXO trunk port models, 1/2/4/8 FXS
ports with lifeline capability.
 Three Gigabit network ports, integrated
PoE+, USB and SD ports.
 Up to 3000 SIP endpoints and up to 200
SIP trunk accounts.
 Up to 10 Video Conference rooms and up
to 80 parties with 1080P HD and up to 300
parties for Audio Conference.
 Zero Configuration endpoint provisioning
and no licensing fees.
UCM630xA Specifications

UCM6300A UCM6302A UCM6304A UCM6308A  Based on Asterisk* Version 16.

 0/2/4/8 FXO trunk port models,
0/2/4/8 FXS ports with lifeline
capability.
 Three Gigabit network ports,
integrated PoE+, USB and SD ports.
 Up to 1500 SIP endpoints and up to
200 SIP trunk accounts.
 Up to 9 Audio meeting rooms with
up to 150 parties.
 Zero Configuration endpoint
provisioning and no licensing fees.
UCM6xxx Specifications
Specifications UCM630x UCM630x Audio

Processor UCM6301 & UCM6302: UCM6300A/6302A:

1x ARM Cortex-A53 Quad-Core 1 x ARM Cortex-A53 Quad-Core
UCM6304 & UCM6308: UCM6304A/6308A:
2x ARM Cortex-A53 Quad-Core 2 x ARM Cortex-A53 Quad-Core
Memory UCM6301 & UCM6302: 2GB UCM6300A & UCM6302A: 2GB
UCM6304 & UCM6308: 4GB UCM6304A & UCM6308A:
2GB (Master) + 1GB Slave memory.
Internal Memory UCM6301 & UCM6302: 32GB UCM6300A & UCM6302A: 32GB
UCM6304 & UCM6308: 128GB UCM6304A & UCM6308A: 32GB
ISDN port No No

FXO ports UCM6301: 1 ; UCM6302: 2 ; UCM6300A: 0 ; UCM6302A: 2 ;

UCM6304: 4 ; UCM6308: 8; UCM6304A: 4 ; UCM6308A: 8;
FXS ports UCM6301: 1 ; UCM6302: 2 ; UCM6300A: 0 ; UCM6302A: 2 ;
UCM6304: 4 ; UCM6308: 8; UCM6304A: 4 ; UCM6308A: 8;
Network ports 3 Gigabit RJ45 ports with 3 Gigabit RJ45 ports with integrated
integrated PoE+ PoE+
USB and SD Card slot Yes Yes
Integrated NAT router Yes Yes
UCM6xxx Specifications
Specification UCM630x UCM630x Audio
Users Up to 3000 (UCM6308) Up to 1500 (UCM6308A)
Concurrent calls UCM6301: Up to 75 ; UCM6300A: Up to 50 ;
UCM6302: Up to 150. UCM6302A: Up to 75.
UCM6304: Up to 300 ; UCM6304A: Up to 150 ;
UCM6308: Up to 450 UCM6308A: Up to 200
SIP Trunks Up to 200 Up to 200
Audio meetings UCM6301: 3 AC with up to 75 parties. UCM6300A: 3 AC with up to 50 parties.
UCM6302: 8 AC with up to 150 parties. UCM6302A: 5 AC with up to 75 parties.
UCM6304: 15 AC with up to 200 parties. UCM6304A: 7 AC with up to 120 parties.
UCM6308: 25 AC with up to 300 parties. UCM6308A: 9 AC with up to 150 parties.
Video meetings UCM6301: 4 VC rooms: up to 20 parties.
UCM6302: 6 VC rooms: up to 30 parties.
N/A
UCM6304: 8 VC rooms: up to 60 parties.
UCM6308: 10 VC rooms: up to 80 parties.
Redundant PSU Only UCM6304 / UCM6308 Only UCM6308A
LED Indicators and - LCD Display is available on all the models. - LCD Display is available on all the models.
Graphic LCD Display - Only UCM6304 / UCM6308 models have LED indicators. - Only UCM6308A models have LED indicators.
Rack mountable - UCM6300A / UCM6302A / UCM6304A:
- UCM6301 / UCM6302: Wall mount and Desktop.
- Wall mount and Desktop.
- UCM6304 / UCM6308: Rack mount and Desktop.
- UCM6308A: Rack mount and Desktop.
Call Management
Call
Transfer
Call SIP
Forward Forking

Call Hold

Call
DND
Routing
Call
Park
Call Management
SIP Forking – Multiple registration per extension

 UCM supports multiple registration per extension. The user can register from multiple devices to UCM
using a single account.

 Up To 10 Concurrent Registrations per extension

GS Wave:
Extension 1000

Register

Home Phone:
Extension 1000

Office Desk Phone:

Extension 1000

17
Call Management
Call Transfer

Call Transfer allows a user to transfer calls to another phone.

Our UCM supports 3 types of Transfer :

 Blind Transfer : Blind Transfer involves passing a call

without notifying the recipient, Music on hold during
transfer process

 Attended Transfer : Attended Transfer involves passing a

call by notifying the recipient, first hold initial call then
call recipient while first call is waiting, then connect the 2
calls.

 Seamless Transfer : Seamless Transfer allows user to

perform blind transfer using UCM feature code without
having music on hold presented during transfer process.
18
Call Management
DND (DND Whitelist)

DND, Do not Disturb allows of an extension to ignore any incoming calls.

This can be done from Both UCM and the client (depending on the client), in addition to that our UCM allows
users to configure extensions/numbers that will be able to reach your extension even if DND is enabled, the
options is Named DND whitelist.

19
Call Management
Call Forward (CFN, CFB, CFU, CF Whitelist)

Call forwarding, or call diversion, is a telephony feature which

redirects a telephone call to another destination, which may be, for
example, a mobile, voicemail box or another phone number where
the desired Called party is available.

UCM supports 3 Types of call forward :

 CFN : Call Forward when No answer

 CFB : Call Forward when Busy
 CFU : Unconditional Call Forward

20
Call Management
Call Hold

Call hold is a key feature of a telephony system. When a call is placed on hold, the
caller is parked and is enable to communicate with the person on the other side of
the line. Music on hold is generally played to the caller. The person who has placed
the caller on hold is able to perform other tasks during the hold time.

UCM users can configure the music on hold class and upload music files through the
web interface, files should be in Wav 8KHz Mono format with maximum size of
30MB

UCM users can also record the music on hold by initiating a recording call to an
extension, answer the call and start the recording. Once you hang up and refresh
the page then the new music on hold file will be available in the list.

21
Call Management
Call Park
This feature allows you to place a call on hold, so it can be retrieved from
another phone in the system.

To use Call Park: if you are on an active call at your phone, you can park the
call in 2 ways:
 Press #72 and the call will be parked.
 Initiate a blind transfer then dial 700 to park the call.

The Parking lot number will be announced after parking the call.

You can retrieve the call in another room or someone else on another phone
in your system by dialing the call park extension number.

If the call is not retrieved after the timeout, it will ring back the phone that
parked it.

22
Call Management
Call routing – Outbound routes

 Dial Pattern
 Secured Route
 Strip/Prepend Digits
 Failover
 Privilege/Permission
 Outbound route per Source
caller ID
 Call Duration Limit
 PIN Group
 Blacklist
 Time condition
 DOD

23
Call Management
Call routing – Inbound routes

 DID Pattern
 Trunk Selection
 Direct Calls to IVR, Extension, Ring
group, Voicemail, FAX
 Scheduled Inbound Routes
 Inbound route per caller ID
 Blacklist
 Multiple modes

24
UC Features
Dial By Call
IVR Speed Dial
Name Recording

Distinctive Busy Camp- Announcem

Conference Ring Tones On ent Center

Presence & Wake up

Voicemail DISA
IM service

Follow me Call Barge Callback Fax

Paging & Pickup

Intercom Group
Ring Group Call Center
UC Features
IVR

IVR (Interactive Voice Response) is a function that greets and guides

callers through a telephony system in a friendly and timely manner
allowing them to reach an endpoint, leave messages or speak to an
operator.

UCM6xxx IVR main features are:

 Record or Upload Custom Voice Prompts (5MB)

 Supports Multiple Languages
 Scheduled Routes to Specific IVR Prompts e.g. Afterhours,
Holidays, Maintenance
 Nested IVRs up to 5 levels

26
UC Features
IVR

Incoming calls to UCM6xxx can be pointed to reach IVR used as auto-attended

(with or without a receptionist) and then customers will be able to select the
destination to reach by a simple key press.

Available options for Key Pressing Events:

 Extension  Fax
 Voicemail  Custom Prompt
 Conference Rooms  Hang-up
 Voicemail Group  DISA
 IVR  Dial By Name
 Ring Group  External Number
 Queues  Callback
 Page Group

27
UC Features
Conference
Conference Bridge feature allows to have several people
participating in call. Different Conference Bridges can be
defined on UCM6XXX and have following main features:

 Public or Private
 Recording Option
 Bridging of Multiple Parties
 Caller Menu
 User Invite
 Caller Announcements

28
UC Features
Voicemail Management

Voicemail is a VoIP Telephony feature, basically, It is a voice message

that a caller leaves when the person called is absent or busy on
another call.

And the called person can later on check the stored voice message by:

 Dialing Voicemail number.

 Receiving voicemail via email
 Accessing user portal

User can listen/delete messages or even change the Voicemail

prompts.

1 hour voicemail occupies around 5MB

30
UC Features
Pickup Group

The UCM supports pickup group feature which allows users

to pick up incoming calls for other extensions if they are in 1 4
the same pickup group, by dialing "Pickup Extension"
feature code (by default *8).
PICKUP GROUP

2
Dial*8

31
UC Features
Paging/Intercom

Paging and Intercom can be used to make an

announcement over the speaker on a group of
phones. Targeted phones will answer immediately
using speaker.

Paging is one-way communication so people can hear Paging/Intercom Group

the message but cannot respond to it.

Intercom is two-way communication so people can

hear the message and also respond back to the person
initiating the call.

The paging and intercom can be used via feature code

to a single extension or a group.

32
UC Features
Call Barge
 Offers ability to listen in on agents calls, whisper to the agent and barge in on both calling parties.
 Integrated with active call status under Web UI  System Status  Active Calls.
 Can be used with Feature Codes.
 Different modes supported:

 Listen
In “Listen” mode, monitor’s extension can hear both parties in the active call but monitor’s audio
won’t be heard by either side of the monitored active call. Default feature code is *54.

 Whisper
In “Whisper” mode, monitor’s extension can hear both parties in the active call but monitor’s audio
can only be heard by the monitored extension. Default feature code is *55.

 Barge
In “Barge” mode, monitor’s extension can talk to both parties in the active call. The call will be
established similar to a three-way conference. Default feature code is *56.
33
UC Features
Speed Dial

Speed dial allows users to call a certain destination by

pressing one or four digits.

Supported destinations are:

 Extension
 Voicemail
 Conference room
 Voicemail group
 IVR
 Ring group
 Call queue
 Paging group
 Fax
 DISA
 Dial by name
 External number

35
UC Features
Ring Group

 A Ring Group is a feature that allows you to have

multiple phones ring when one extension or number
is dialed. It’s a great way for a business to share the
distribution of incoming calls among employees. It is
often used to efficiently distribute calls to specific
departments such as Sales, Customer Support and
Accounting. RING GROUP
 Up to 100 members per Group
 Parallel or Serial ringing mode
 Custom Prompt
 Auto record
 Default Destination
36
UC Features
Call Center
UCM supports light weight call center features including
virtual queue, switchboard and position announcement,
allowing the callers to know their position on the call
queue and giving them the option to either stay on the
line waiting for their turn or activate a callback which will
be initiated by the UCM one an agent is free.

 Virtual Queue.

 Call Queue position announcement

 Call Queue Statistics

 Switchboard for all agents

37
UC Features
Call Center/Call Queue – Call distribution Strategy

UCM supports automatic call distribution with many

different strategies that can be set under basic options of
each call queue separately.

 Ring All
 Fewest Calls
Ring all available agents simultaneously until one answers.
The system assigns the call to the agent with the lowest
 Linear number of calls answered so far.
The calls will ring at agents’ phones following an order
 Random
previously configured. Only for static agents.
Ring a random agent.
 Least Recent
 Round Robin
The system selects the agents who have not answered the
Ring the agents in Round Robin scheduling with memory.
phone for the longest period.
38
UC Features
Call Center – Call Queues Agents

Call queue Agents are members defined to answer the queue

calls. Agents can be either static or dynamic.

UCM Model Max Agents in Call Queue

UCM6301 25
UCM6302 50
UCM6304 80
UCM6308 160

39
UC Features
Fax Service

 The UCM6xxx supports T.30/T.38 Fax and Fax No Fax Machine? No Problem!
Pass-through. It can convert the received Fax
to PDF format and send it to the configured
Email address.

 The list of received Fax files will be displayed

in the same web page for users to view,
retrieve and delete.

 Sending fax can be done either by using a Fax

machine connected to one of the FXS ports
or by uploading the PDF file through the user
portal.

 Fax/T.38 settings can be accessed via Web

GUI  Call Features  FAX/T.38 47
UC Features
Call Recording

 55 MB usage for about 1 hour call recording.

 Recording length can be extended via USB flash

drive

 Recording files are wav format and can be

played and downloaded from UCM6XXX series
web GUI

 Automatic recording can enable for specific

extensions, ring groups, call queues, conference
rooms, trunks.

49
Grandstream High Availability Solution for UCM6300 Series
Introduction

 Grandstream High Availability is a solution that provides

continuous availability for any UCM server deployed for
critical missions and for applications and services that
need to run with a minimal downtime.

 It uses redundant software performing a similar function

using two UCM servers, one of them will be the Active
UCM and the other will be the Passive.
UCM630x / UCM630xA HA SYSTEM
UCM630x / UCM630xA HA Setup
• The two UCMs must be the same model and firmware
version.
• The data on the primary UCM will be synchronized to
the secondary UCM in real time manner and the
secondary UCM monitors primary UCM’s status
regularly.
• When the primary UCM runs into hardware or severe
software issues, the secondary UCM will take over
immediately and goes in “active” status, becoming the
primary server.

• Heartbeat connection between the two UCMs

• UCMs must be in Route OR Switch network mode

• The IPv4 address must be using static IP.

DEVICE
MANAGEMENT
Deploy and Manage
Grandstream Solutions
Centrally
Grandstream complements its
hardware solutions with centralized
deployment, management, and
troubleshooting systems
GDMS
Grandstream Device Management System

The Grandstream Device Management System (GDMS) is a zero-touch cloud provisioning and
management system that provides easy-to-use enterprise-level tools to manage Grandstream
products before, during, and after deployment. It provides a centralized interface to provision,
configure, monitor, and troubleshoot.

• Configuration, management, and monitoring of

devices, accounts, and firmware
• Real-time network monitoring; analytical system
reports
• Can manage batches of devices by deployment sites,
device groups and models within those sites
• Intelligent alarms, troubleshooting, and diagnostic
reports
• Automate firmware updates, device resets, and
use-logs
GRP Series
IP Phones
Carrier-Grade IP Phones

Next-generation IP Phones designed for

mass deployment. This series features a
sleek new design, a reimagined user
experience, and many other advanced
features.
Definition of Terms
SIP Lines
# of concurrent calls that a device can handle
# of call appearances

SIP Accounts
Extension / Local number

SIP (Session Initiation Protocol)

Protocol used as Signaling by Grandstream devices.

BLF (Busy Lamp Field)

Presence indicator that allows you to see who in your organization is available (or not) for a
phone call at any given time.

63
GRP2601/P – Essential IP Phone NEW
2-Line Essential IP Phone

• 2 SIP accounts, 2 lines

• GDMS online device management and
provisioning
• 3-way audio conferencing for easy conference
calls
• Equipped with noise shield technology to
minimize background noise
• Electronic Hook Switch (EHS) support for
Plantronics, Jabra, and Sennheiser headsets
• GRP2601P includes built-in PoE to power the
device and provide a network connection
GRP2602/P – Essential IP Phone NEW
2-Line Essential IP Phone

• 2 lines, 4 SIP accounts

• Supported by GDMS which provides a centralized
interface to configure, provision, manage, and monitor
Grandstream devices

• EHS support
• Full-duplex speakerphone with HD audio
• Backlit LCD
• Noise shield technology to minimize background noise
• GRP2602P includes built-in PoE to power the device
and provide a network connection

• Swappable faceplate for easy logo customization

GRP2603/P – Essential IP Phone NEW
3-Line Essential IP Phone

• 3 lines, 6 SIP accounts, up to 3 call appearances

• Supported by GDMS
• EHS Support
• Full-duplex speakerphone with HD audio
• Noise shield technology
• Dual switched auto-sensing 10/100/1000 Mbps
Gigabit Ethernet ports with integrated PoE
(GRP2603P only)
• Supports 4-way audio conferencing
• Swappable faceplate for easy logo customization
GRP2604/P – Essential IP Phone NEW
3-Line Essential IP Phone

• 3 lines, 6 SIP accounts, up to 3 call appearances

• Supported by GDMS
• EHS Support
• Full-duplex speakerphone with HD audio
• Noise shield technology
• Dual switched auto-sensing 10/100/1000 Mbps
Gigabit Ethernet ports with integrated PoE
(GRP2604P only)
• Supports 4-way audio conferencing
• Swappable faceplate for easy logo customization
• 10 Customizable BLF Keys
GRP2612/W/P
Carrier-Grade IP Phone 4 Pages LCD
• 2.4” color LCD
• 2 SIP accounts and 4 multi-purpose line keys
• Up to 16 virtual multi-purpose keys
• Dual switched auto-sensing 10/100 Mbps
Ethernet ports
• HD audio and OPUS support
• Enterprise-level protection including secure
boot, dual firmware images, and encrypted
data storage

• Wi-Fi option with GRP2612W

• PoE option with GRP2612P
GRP2613
Carrier-Grade IP Phone

• 2.4” color LCD

• 3 SIP accounts and 6 multi-purpose line
keys
• Up to 24 virtual multi-purpose keys
• Dual switched auto-sensing 10/100/1000
Mbps Gigabit ethernet ports with PoE

• HD audio and OPUS support

• Enterprise-level protection including
secure boot, dual firmware images, and
encrypted data storage
GRP2614
Carrier-Grade IP Phone with Dual LCD

• 2.8” color LCD

• 2.4” secondary color LCD for digital BLF/SD
• 4 SIP accounts and 4 multi-purpose line keys
• Dual switched auto-sensing 10/100/1000
Mbps gigabit Ethernet ports with PoE

• HD audio and OPUS support

• Integrated dual-band Wi-Fi
• Enterprise-level protection including secure
boot, dual firmware images, and encrypted
data storage
GRP2615
Carrier-Grade IP Phone
• 4.3” color LCD
• 5 SIP accounts and 10 multi-purpose line keys
• Up to 40 virtual multi-purpose keys
• Dual switched auto-sensing 10/100/1000 Mbps
Gigabit ethernet ports with PoE

• HD audio and OPUS support

• Integrated dual-band Wi-Fi
• Extension module support (GBX20)
• Enterprise-level protection including secure
boot, dual firmware images, and encrypted
data storage
GBX20
Extension Module

• Used with the GRP2615, GRP2624, and

GXV3350
• 20 per page (each module contains 2 pages, for
up to 40 lines per module, up to 160 with 4 daisy-
chained modules)

• 4.3 inch (272x480) TFT color LCD

• Powered by host phone; when 2 or more
GBX20 are connected, the included 12V/1A
PSU will power them
GRP2616
Carrier-Grade IP Phone

• 4.3” (480 x 272) color LCD,

• 2.4” (240 x 320) secondary LCD
• 6 SIP accounts and 6 lines
• 48 multi-purpose keys
• Dual switched auto-sensing 10/100/1000 Mbps
Gigabit ethernet ports with PoE

• HD audio
• Integrated dual-band Wi-Fi
• Extension module support
• Enterprise-level protection including secure boot,
dual firmware images, and encrypted data storage
GXV Series IP
Video Phones
Innovation and High-End
A/V Capabilities
These Grandstream devices provide
the power of video communications
with integrated web features and the
functionality of a tablet.
GXV3350
High-End Smart Video Phone for Android

• 16 lines with up to 16 SIP accounts supported

• Built-in 1-megapixel camera for video calling with
privacy shutter

• Runs on Android 7.0 operating system

• Built-in Bluetooth and dual-band Wi-Fi
• Dual-switched auto-sensing Gigabit ports with built- in
PoE/PoE+
• Dual-mic HD speakerphone with advanced echo
cancellation
• Extension module support (GBX20)

• 5” capacitive 5-point touch screen

• 6-way HD audio conferencing & 3-way 720p 30fps
HD video conferencing capability
GXV3370
Android Video IP Phone

• Runs on Android 7.0

• 7” touch screen
• 16 lines, 7-way audio conference and no cost
3-way video conference
• 3rd party Android™ applications include
Microsoft Lync, Skype, Twitter, Facebook

• Wi-Fi, PoE, Bluetooth, USB, SD, mini HDMI, EHS

• 4-core processor and 2 GB RAM and 8 GB
eMMC Flash for maximum performance
GXV3380
High-End Smart Video Phone for Android

• 16 lines with up to 16 SIP accounts supported

• Built-in 2 megapixel camera for video calling with
privacy shutter

• Runs on Android 7.1.1 operating system

• Built-in Bluetooth and dual-band Wi-Fi
• Dual-switched auto-sensing Gigabit ports with built-in
PoE/PoE+
• Dual-mic HD speakerphone with advanced echo
cancellation
• 8” capacitive 10-point touch screen
• 7-way HD audio conferencing & 3-way 1080p 30fps
HD video capability
WIRELESS
IP PHONES
Mobility andAdvanced
Features for All
Mobilize your network with wireless
phone solutions from Grandstream.
These phones are an excellent option
for companies, warehouses, retail
stores, and more!
DP730
High-End DECT Wireless IP Phone

• 2.4” color LCD screen

• Up to 10 lines and SIP accounts per handset

• HD audio, integrated 3.5mm Headset Jack, Speakerphone

• Up to 40-Hour Talk Time, 500-Hour Standby Time

• Powerful antenna supports 400 meters outdoors and 50

meters indoors.

• Advanced telephony features such as hold, transfer,

forward, call parking, music on hold, and more

• Three dedicated softkeys, two volume keys, one

programmable push to talk key.
DP722
Mid-Tier DECT Wireless IP Phone

• 1.8” color LCD screen

• Up to 10 lines and SIP accounts per handset

• HD audio, integrated 3.5mm Headset Jack, Speakerphone

• Up to 20 Hour Talk Time, 250-Hour Standby Time

• Antenna supports 350 meters outdoors and 50 meters

indoors.

• Advanced telephony features such as hold, transfer, forward, call

parking, music on hold, and more

• Two configurable softkeys, push-to-talk support

DP752
DECT IP Base
Station
• Up to 10 SIP accounts per system

• Up to 5 DP DECT handsets per base station

• Integrated PoE

• 3-way conferencing

• Supports DP722 and DP752 push-to-talk functionality

• 400m range outdoors, 50m indoors

• TLS/SRTP/HTTPS encryption security

WP820
Enterprise Portable Wi-Fi Phone

• Dual-band Wi-Fi with advanced roaming support

• HD Voice and dual MIC design with AEC and noise
shield technology
• 2 SIP Accounts, 2 Lines
• Android 7.0 operating system allows customized
android apps for the WP820’s screen/keyboard

• Built in accelerometer, panic button, push-to-talk

• 7.5 hour talk time, 150 hour standby time
WP810
Cordless Wi-Fi IP Phone

• Dual-band Wi-Fi with advanced roaming support

• HD Voice and dual MIC design with AEC and noise shield
technology

• 2 SIP Accounts, 2 Lines

• Rechargeable 1500mAh battery, 6-hour talk time, 120-hour
standby

• Configurable button for push-to-talk

• Micro USB port and 3.5mm headset jack
GRP2601P

86
UCM Deployment with Analog Trunk, IP Phones, and Analog Phones

GXW4232 GRP2601P

87
88
GXW4232 GRP2601P

89
GRP2601P

90
91