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The GXP1625 is Grandstream’s standard Basic IP phone. This 2-line IP Phone includes 3-way conferencing to keep workers in-touch and productive. A 132x48 backlit LCD screen creates a clear display for easy viewing. Additional features such as dual-switched 10/100mbps ports, HD audio, multi-language support, integrated PoE (GXP1625 only) and 3 XML programmable soft keys allow the GXP1625 to be a high quality, versatile and dependable phone.
A reliable Basic IP phone, the GXP1625 delivers a suite of powerful VoIP features in an easy-to-use fashion with a modern design. The GXP1625 offers support for up to 2 lines and features full HD audio on both the speaker and handset to allow users to communicate with the utmost clarity. The GXP1625 features built-in Poe. Additional features include 3-way conferencing to maximize productivity, 3 XML programmable soft keys for customization and EHS support for Plantronics headsets. These features allow the GXP1625 to be a high-quality and easy-to-use Basic IP phone.
As all Grandstream IP phones do, the GXP1625 features state-of-the-art security encryption technology (SRTP and TLS). The GXP1625 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.
- 2 SIP accounts, 2 line keys, 3-way conferencing,
- 3 XML programmable context-sensitive soft keys
- Dual-switched 10/100 mbps ports, integrated PoE on GXP1625
- HD audio on speakerphone and handset
- EHS support for Plantronics headsets
- Up to 500 contacts, call history up to 200 records
- Power supply included in the box
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