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Home/VoIP Hardware/VoIP Phones/HTEK VoIP Phones/HTEK UC912G

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HTEK UC912G

€83.00

(€98.77 inc tax)

Return period: 10 days

CODE: HTEKUC912G

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Description
Technical data
Features
System Requirements
Portsip Soft & Hardware Requirements
Portsip Features
Agenda

UC912G delivers superior HD audio quality, rich and leading edge telephony features, all the functions of software are the same as other UC900 series models. It is a perfect choice for small to medium businesses looking for a high quality, feature rich IP Phone at an affordable cost.

Phone Features: Up to 4 SIP accounts

Call hold, mute, DND

One-touch speed dial, hotline

Call forward, call waiting, call transfer

Redial, call return, auto answer, direct IP call

5-way conference

Group listening, SMS, emergency call

Ring tone selection/provisioning

Set date time automatically or manually

Dial plan per account

RTCP-XR (RFC3611), VQ-RTCPXR (RFC6035)

Action URL/URI

Voice Codecs Features: HD voice, HD handset, HD speaker

Codecs: iLBC, G.722, G.711(A/μ), Opus

GSM_FR, G.723, G.729AB, G.726-32

DTMF: In-band, RFC 2833, SIP INFO

Full-duplex hands-free speakerphone with AEC

VAD, AGC, CNG, AEC, PLC, AJB

Directory: Local phonebook up to 1000 entries

XML/LDAP remote phonebook

Intelligent search method

Phonebook search/import/export

Call history: dialed/received/missed/forwarded

Black list

IP-PBX Features: Busy Lamp Field (BLF), bridged Line Appearance(BLA)

Anonymous call, anonymous call rejection

Message Waiting Indicator (MWI) , voice mail

Call park, call pickup

Intercom, paging

Music on hold

Hot-desking

Display and Indicator: 192x64 pixel graphical LCD with backlight

LED for call and message waiting indication

Illuminated LEDs for line status information

Intuitive user interface with icons and soft keys

National language selection

Caller ID with name, number

Feature keys: 4 line keys with LED

4 line keys can be programmed up to

12 various features (4-page view)

Illuminated speaker key/headset key/mute key

4 context-sensitive “soft” keys

6 navigation keys, volume control keys

Illuminated speaker key/headset key/mute key 6 features keys: voice mail, headset, mute,speaker, transfer, redial

Interface: Dual-port Gigabit Ethernet

Power over Ethernet (IEEE 802.3af), class 2

1xRJ9 (4P4C) handset port

1xRJ9 (4P4C) headset port

Physical Features: Stand with 2 adjustable angles

Wall mountable

External universal AC adapter : AC 100~240V input and DC 5V/1.2A output

Power consumption (PSU): 1.8~4.4W

Power consumption (PoE): 2.3~5.3W

Operating humidity: 10~95%

Operating temperature: -10~50°C

Management: Configuration: browser/LCD-Menu/auto-provision

Auto provision via HTTP/HTTPS FTP/TFTP

Auto-provision with PnP

Reset to factory, restart, reboot

Local tracing log export, system log

Phone lock for personal privacy protection

Network and Security: SIP v1 (RFC2543), v2 (RFC3261)

SIP server/proxy redundancy

NAT Traversal: STUN mode

DHCP/static/PPPoE/IEEE802.1X/open VPN

HTTP/HTTPS web server

Time and date synchronization by SNTP

DNS-NAPTR/DNS-SRV(RFC 3263)

QoS: 802.1p/Q tagging (VLAN), Layer 3 ToS DSCP

TLS(Transport Layer Security), SRTP

HTTPS certificate manager

AES encryption for configuration file

Digest authentication using MD5/MD5-sess

IPv6 supported

Packing: Qty/CTN: 10 PCS

Phone Size:184MM(L)*186MM(W)

N.W/CTN: 10.14 KG

Phonebox Size:200MM(L)*193MM(W)*102MM(H)

G.W/CTN: 11.14 KG

Carton Meas: 530MM(L)*420MM(W)*220MM(H)

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

HTEK UC912G

PortSIP PBX Features

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