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A standard Basic IP desktop phone to keep workers in-touch and productive. The GXP1620 features effective and essential functionalities to create an easy-to-use experience for a user with light to medium call volume. Its focus on essential features and standard call support makes the GXP1620 a versatile and dependable phone.
The GXP1620 is a reliable Basic IP phone for the user who requires standard features for a light to medium call volume. Stay in touch with others with its 2 lines/SIP accounts, enjoy crystal clear HD audio and utilize its dual-switched 10/100 mbps ports for a flexible deployment. Maximize your productivity with essential supported features such as 3-way conferencing and 3 XML programmable soft keys. Together, these featuresmake the GXP1620 an easy-to-use and effective Basic IP phone.
As all Grandstream IP phones do, the GXP1620 features state-of-the-art security encryption technology (SRTP and TLS). The GXP1620 supports a variety of automated provisioning options, including zero-configuration with Grandstream’s UCM series IP PBXs, encrypted XML files and TR-069, to make mass deployment extremely easy.
Features:
- 2 SIP accounts, 2 line keys, 3-way conferencing,
- 3 XML programmable context-sensitive soft keys
- Dual-switched 10/100 mbps ports,
- HD audio on speakerphone and handset
- EHS support for Plantronics headsets
- Up to 500 contacts, call history up to 200 records
Technical data
Features
System Requirements
A Windows desktop or Windows server OS: Windows 7/8, Windows 10, Windows Server 2008 R2 with SP1, 2012 R2, 2016 R2) IPv4/IPv6 Windows firewall
This document assumes that the Windows OS is already deployed and administrators of PortSIP PBX are allocated the administrator permission to Windows.
Hardware and Software Dependencies
OS Supported by PortSIP PBX includes:
Linux Server:
CentOS 7 or higher, 64bit; gcc/g++ 6.4 or higher
Ubuntu 16.04.4 or higher, 64bit; gcc/g++ 6.4 or higher
Debian 9.0 or higher, 64bit; gcc/g++ 6.4 or higher
Windows Desktop:
Windows 7, 8 and 10, 64-bit
Windows Server:
Windows 2008 R2 SP1, 2012 R2, 2016 R2, 64-bit
Important: The OS must be up to date.
Cloud and Virtualization Environment Supported
To build high-availability communication solution to help clients reduce cost and improve communication performance, PortSIP PBX commits support on cloud services and have confirmed availability on following cloud and virtualization environment:
- VMware ESX 5.X and above.
- Linux HyperV
- Microsoft HyperV 2008 R2 and above
- Amazon AWS
- UCloud
- Alibaba Cloud
- Linode
- Digital Ocean
- Godaddy VPS and Cloud
- Tencent Cloud
System performance depends on following key factors:
- Maximum simultaneous calls needed for PBX
- Maximum online users needed for PBX
- Recordings for calls
- Record audio only or both of audio, video
- Maximum online users for audio/video conferences on PBX
- Maximum IVR (Virtual Receptionist) on PBX
- Maximum Call Queues on PBX
- Maximum Ring Groups on PBX
Depending on the key features listed above, PortSIP PBX is able to run on PCs and servers with various CPSs ranging from Intel i3 CPU to Inter Xeon
Other Requirements
- Latest Firefox, Google Chrome or Internet Explorer
- Microsoft .NET Framework version 4.5 or higher
- Knowledge of Linux and Linux Internet administration
- Knowledge of Windows and Windows Internet administration
- A constant internet connection to service.portsip.com on port 6881.
- A constant internet connection to stun.portsip.com and stun1.portsip.com on port 3478.
- A constant internet connection to stun4.l.google.com on port 3478.
PortSIP PBX Features
PortSIP PBX free Edition and Full Edition are equipped with same features, with the only difference that the free Edition only support up to 3 simultaneous calls only.
| Features | Free Edition | Full Edition |
| Extensions | Unlimited | Unlimited |
| Number of Simultaneous Calls Supported | 3 | >10,000 |
| Linux Support (CentOS, RHEL, Debian, Ubuntu) | Y | Y |
| Multi-Tenant | Y | Y |
| Call Logging | Y | Y |
| Call Forward on Busy or No Answer | Y | Y |
| Call Routing by DID | Y | Y |
| Auto Attendant / Digital Receptionist | Y | Y |
| Voicemail/ Music on Hold | Y | Y |
| Central Phonebook | TBD | TBD |
| Call Transfer | Y | Y |
| MWI – Message Waiting Indicator | Y | Y |
| Ring Extension & Mobile Simultaneously | Y | Y |
| Automatic Pickup on Busy | Y | Y |
| Supports SIP Trunks/ Gateways | Y | Y |
| Custom SMTP Server | Y | Y |
| Custom FQDN | Y | Y |
| Busy Lamp Field (BLF) | Y | Y |
| Call Reporting | Y | Y |
| Call Parking / Pickup | Y | Y |
| Call Queuing | Y | Y |
| Audio Call Recording | Y | Y |
| Video Call Recording | Y | Y |
| Intercom/ Paging | Y | Y |
| Call Recordings Management | Y | Y |
| Configure BLF’s from the Clients | Y | Y |
| Web-based Management Console | Y | Y |
| Automated Provisioning of Devices | Y | Y |
| Real Time Web-based System Status | Y | Y |
| Integrated Web Server | Y | Y |
| Easy Backup and Restore | Y | Y |
| VMware / Hyper-V Compatibility | Y | Y |
| Scheduled Backup | Y | Y |
| Scheduled Restore | Y | Y |
| Inbuilt Fail Over Functionality | Y | Y |
| Standby License | Y | Y |
| Deploy as cluster | Y | Y |
| See the Presence of Your Colleagues | Y | Y |
| Receive Voice Mail via Email | Y | Y |
| Advanced Forwarding Rules | Y | Y |
| Setting Up Conference Calls | Y | Y |
| Click2Call | Y | Y |
| View Presence of Remote Offices | Y | Y |
| Advanced Queue Strategies | Y | Y |
| Advanced Call Reporting | Y | Y |
| Real Time Queue Statistics | Y | Y |
| Ability to Use PortSIP VoIP SDK | Y | Y |
| Real Time Queue Monitoring | Y | Y |
| Call Recordings Search | Y | Y |
| Supports External Agents | Y | Y |
| Android Client (Provide rebrand/OEM) | Y | Y |
| iOS Client (Provide rebrand/OEM) | Y | Y |
| Windows Client | Y | Y |
| Mac Client | Y | Y |
| Web client | Y | Y |
| Mobile PUSH | Y | Y |
| Avoid NAT Problems | Y | Y |
| Automatic Plug & Play Phone Provisioning | Y | Y |
| Manage IP Phones Network Wide from Console | Y | Y |
| Restart Phones Remotely | Y | Y |
| Supports Popular SIP Phones | Y | Y |
| Provide Client VoIP SDK(Android, iOS, Windows, macOS) | Y | Y |
| Plugin Free – WebRTC | Y | Y |
| One-click conference | Y | Y |
| Meeting Recording | Y | Y |
| Unlimited Users | Y | Y |
| Meeting Participants Included | 3 | 100 |
| Full REST API | Y | Y |
| Security | Y | Y |
| E164 Number processing | Y | Y |
Agenda