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HTEK UC912P

€70.00
(€83.30 inc tax)
10 days
HTEKUC912
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UC912 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)

 

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