Electronics

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Electronics Suppliers and Manufacturers

Telecommunication

elec

Find Telecommunication from the best B2B online marketplace. Find a reliable Telecommunication supplier, vendor, manufacturer or distributor to get free quote on HKTDC Sourcing.

Telecommunication Suppliers and Manufacturers

Linkco Int'l Ltd

   Bluetooth V3.0+EDR, class II (Broadcom 20771)   Up to 15 meters in working distance (open space)   Bluetooth profiles: A2DP1.2/AVRCP1.0/HSP/HFP   Up to 14 hours talk time   Up to 10 hours music playback time   Up to 6 days standby time   Voice recognition for true hands free by saying 'Answer' to answer the call and 'Ignore' to reject the call.   Voice prompt in 5 languages: English, Spanish, German, French, Chinese   Multipoint: Supports 2 phones connection   Battery meter on iPhone   Support 3.5mm line-in for use in case of out of power.   Support VOIP, connect with computer for online chat.   Vice dialing/ redialing   Product dimension: D4.0 mm x 1.5 mm   Weight: 120 g   Colors: black, white, hot pink  

Headphone

Avantec Mfg Ltd

General
-&nbsp;VoIP (SIP) Business Phone
-&nbsp; Maximum of 4 VoIP accounts
-&nbsp; 128 x 64 dot matrix LCD with greyish backlight
-&nbsp; Wideband G.722 Codec
-&nbsp; IPv4 / IPv6
-&nbsp; WAN port and LAN port
-&nbsp; PoE (This option must be specified at the time of order placement)
Server Compliance
-&nbsp; Asterisk / Broadsoft / 3CX / Alcatel / NEC / Avaya / Aastra / Mitel
Call features
-&nbsp; Call Hold
-&nbsp; Call Flash
-&nbsp; Call Forward
-&nbsp; Call Waiting Indication
-&nbsp; Call transfer (blind/attended)
-&nbsp; Caller ID display
- &nbsp;Redial List and redial
-&nbsp; Do not disturb DND
-&nbsp; Mute-&nbsp; Auto-answer
-&nbsp; Auto-complete dialling
-&nbsp; Smart redialling
-&nbsp; 3-way conferencing
-&nbsp; Configurable hotkeys for speedial / voicemail
-&nbsp; Message Waiting Indication (MWI) LED
-&nbsp; Volume control
-&nbsp; Direct IP call without SIP proxy
-&nbsp; Ring tone selection
-&nbsp; Phonebook (15+ entries), Black list
-&nbsp; Call history: dialled/received/missed/forwarded
-&nbsp; LCD menu user interface
-&nbsp; Web Portal for administrators
Physical Features
-&nbsp; 128 x 64 dot matrix display
-&nbsp; Four softkeys
-&nbsp; Eight function keys including one programmable memory button
-&nbsp; LED for MWI
-&nbsp; RJ45 10/100M Ethernet WAN port
n Gigabit Ethernet WAN port (August 2015)
-&nbsp; RJ45 10/100M Ethernet LAN port
n Gigabit Ethernet LAN port (August 2015)
-&nbsp; Power adapter: AC 100~240V input DC 12V/1.2A output (This option must be specified at
the time of order placement)
-&nbsp; Power over Ethernet IEEE 802.3af (This option must be specified at the time of order
placement)
Codecs and Voice Features
-&nbsp; Wideband codec: G.722
-&nbsp; Narrowband codec: G.711&micro;/A, ILBC
-&nbsp; G.729
-&nbsp; VAD, CNG, AEC, PLC, AJB, AGC
-&nbsp; Full-duplex speakerphone with AEC
Network Features
-&nbsp; SIP v1 (RFC2543), v2 (RFC3261)
-&nbsp; IPv4 / IPv6
-&nbsp; VLAN QoS (802.1pq)
-&nbsp; NAT Traversal: STUN mode
-&nbsp; DTMF: In-Band, RFC2833, SIP Info
-&nbsp; Proxy mode and peer-to-peer SIP link mode
-&nbsp; IP Assignment: Static/DHCP/PPPoE
-&nbsp; Bridge/router mode for PC port
-&nbsp; TFTP/DHCP/PPPoE client
-&nbsp; Telnet/HTTP server
-&nbsp; DNS client
-&nbsp; NAT/DHCP server
Management and Provisioning
-&nbsp; F/W provision via TFTP/HTTP
-&nbsp; Auto-provisioning: Option66
-&nbsp; Mass-configuration tool for system integration (check with your local supplier / distributor)
-&nbsp; Telnet

VoIP (SIP) Business Phone

This is a VoIP handset set that comes with one base and six handsets. Using the&nbsp;advanced DECT technology, this handset set supports IPv4 and PoE. This professional VoIP handset set is designed with an LCD screen with backlight for easy reading. 	 For business and home use. 	 DECT, GAP, IPv4 and PoE. 	 1 base&nbsp;and 6 handsets. 	 Colors: white gloss, black gloss, red gloss and&nbsp;yellow gloss. 	 Lighted keypad. 	 Mono LCD with backlight.  

VoIP DECT Handset Set

 	 Maximum of 6 VoIP accounts - IPv4 - WAN port and LAN port - POE - Wideband G.722 Codec 	 Server Compliance Asterisk / Broadsoft / Alcatel / NEC / Avaya / Aastra&nbsp; 	 Dual power supplied Support 1.8GHz (EU) and DECT 6.0 (US) frequency band  

VoIP Dect Phone

&bull; 3&quot; monochrome LCD
&bull; Multi-sip account
&bull; PoE 802.11af
&bull; VLAN, SRTP, HTTPS
&bull; 8 one-touch speed-dial keys
&nbsp;

VoIP Phone

General 	 SIP Accounts : 3 	 Wideband G.722 Codec 	 Narrowband Codec : G.729, iLBC 	 IPv4  
Server Compliance 	 Asterisk&nbsp;/&nbsp;Broadsoft&nbsp; /&nbsp;Alcatel&nbsp;/&nbsp;NEC&nbsp;/ Avaya basic call features supported  Call Features 	 Call Hold 	 Call Mute&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 	 Off-hook Emergency Call 	 Headset : RJ9 	 Hands-free  
Physical Features 	 Trimline form factor 	 Magnetic Hook Switch 	 Wall Mountable&nbsp;&nbsp; 	 RJ45 10/100M Ethernet WAN port 	 7 function keys 	 Hotline 	 line keys (can be defined as function keys at the time of purchase) 	 LCD Soft Keys : 7 	 memory button 	 Power over Ethernet 802.11AF 	 LCD Backlint  Codecs and Voice Features 	 Wideband G.722 Codec 	 Narrowband Codec : G.729, iLBC&nbsp;&nbsp; 	 VAD, CNG, AEC, PLC, AJB, AGC&nbsp;&nbsp;&nbsp; 	 Full-duplex speakerphone with AEC  
Other Features 	 Phonebook : CVS, XML, LDAP, VCF 	 Intercom 	 Paging : By KTS-100 Support 	 Call Park : By KTS-100 Support 	 Call Pick up : By KTS-100 Support 	 Follow-me : By KTS-100 Support 	 Do-not-Disturb : Support 	 Redial List 	 System Log 	 Call Log 	 Dist. Ringtone 	 Ringtone : WAV  

Color LCD Business VoIP Phone

Shenzhen Daerxin Technology Co Ltd

This high-quality&nbsp;call centre phone delivers superb sound quality and receives&nbsp;voices clearly.&nbsp;The phone&nbsp;is sturdily constructed for extended durability.&nbsp;It comes with a&nbsp;well-engineered headset&nbsp;that is comfortable to wear.

 
  Headset for&nbsp;handsfree dialing. 
  Durable construction. 
 

Call Centre Phone

eSky Multimedia Sdn. Bhd.

With eTel mobile VoIP calling card, you will be given able to reach clients and partners around the world easily. You can also call to any regular phone from only 1 c only. You can save up to 95%. We are looking for international reseller to be part of our partner to resell this service.
 
  Virtual pin or physical cards is available. 
  Please refer to http://www.etel888.com for detail. 
 

Mobile VoIP International Calling Card

CEM Solutions Pvt Ltd.

   Stand Alone, powerful voice gateway supports up to 50 simultaneous calls on PSTN/IP link and 5 VoIP registration &amp; VoIP Trunks. No additional licenses required for codec transcoding or web interface   Our Scalable VoIP PRI Gateways ranging from 1 to 2 PRI ports developed to enable direct connection of VoIP traffic to the PSTN/fixed line ISDN for extending communications and reducing call charges  Key Features 
   E1/T1 PRI Gateway; up to 2 SPAN interface   Built In Firewall   Flexible Call Routing   Transcoding (Voice coding G.711 A-law/u-law, G.722 G.729AB) licenses included.   Fax Pass-through.   Stand Alone with Web User Interface.   128 ms DSP based Echo Cancellation  Additional Features 
   Up to 50 simultaneous VoIP to ISDN PRI calls   2 T1/E1/J1 PRI ports with programmable NET/CPE modes   Support SIP v2 (RFC 3261 complaint)   Register with multiple SIP service providers   ISDN network clock synchronization for error-free communication   Interoperable with all the major-brand soft switches and IP-PBXs   PIN authentication for accessing Trunks to prevent unauthorized usage   Call detail records   Web-based configuration and management  

ISDN PRI VoIP Gateway

YEASTAR TECHNOLOGY CO., LIMITED

 	 NeoGate TE100 is a single port VoIP E1/T1/J1 gateway (VoIP to E1/T1/J1, and E1/T1/J1 to VoIP) that supports up to 30 concurrent calls. It&rsquo;s designed to bridge the gap between E1/T1/J1 and VoIP networks. TE100 offers SMBs cost effective additions to an legacy telephone system to bring the true benefits of VoIP. Integrating TE100 into an existing network will allow for inexpensive communication via SIP trunking. Also, it could connect VoIP systems with E1/T1/J1 service from legacy carriers  Specification: 	 E1/T1/J1 Port: 1 (Support PRI, MFC R2, SS7) 	 Protocol: SIP (RFC3261) 	 Transport Protocol: UDP, TCP, TLS, SRTP 	 DTMF: RFC2833, SIP INFO, In-band 	 Codec: G.711 a/u-law, G.722, G.726, G.729 A, GSM 	 LAN: 10/100 Mbps 	 WAN: 10/100 Mbps 	 Size: 213 x 160 x 44mm 	 Power Supply: AC 100~240V, 50~60Hz (DC 12V, 1A)  Features: 	 Trunk Support 	 Call Routing Rules 	 Automatic appending and stripping of digits to dialed numbers 	 Caller ID name and number support 	 FAX support 	 Blacklist 	 Firewall 	 DDNS Support 	 Backup And Restore 	 Easy to install  Benefits: 	 Cost-effective Call Routing 	 Easy-to-navigate GUI 	 Simple Installation 	 Easy to integrate 	 Low power consumption for your green office  Applications: 	 Connect legacy PBX systems to VoIP services 	 Connect legacy PBX systems to remote sites over private VoIP links 	 Connect IP PBX systems to legacy TDM services 	 Phased transition from Legacy PBX to IP PBX  

VoIP-PRI Gateway (VoIP-E1/T1/J1)

    NeoGate TE200 is a dual port VoIP E1/ T1 /J1 gateway (VoIP to E1/ T1 /J1, and E1/ T1 /J1 to VoIP) that supports up to 60 concurrent calls. It&rsquo;s designed to bridge the gap between E1/ T1 /J1 and VoIP networks. TE200 offers SMBs cost effective additions to an legacy telephone system to bring the true benefits of VoIP. Integrating TE200 into an existing network will allow for inexpensive communication via SIP trunking. Also, it could connect VoIP systems with E1/T1/J1 service from legacy carriers.   Specification:   E1/T1/J1 Port:&nbsp; 2 (Support PRI, MFC R2, SS7)    Protocol: &nbsp;SIP (RFC3261)    Transport Protocol:&nbsp; UDP, TCP, TLS, SRTP    DTMF:&nbsp; RFC2833, SIP INFO, In-band    Codec:&nbsp; G.711 a/u-law, G.722, G.726, G.729 A, GSM, &nbsp; ADPCM, Speex     LAN:&nbsp; 10/100/1000 Mbps     WAN:&nbsp; 10/100/1000 Mbps    Size: &nbsp;213x160x44mm    Power Supply: &nbsp;AC 100~240V,50~60Hz (DC 12V, 1A)   Features:   Trunk Support   Call Routing Rules   Automatic appending and stripping of digits to dialed numbers   Caller ID name and number support   FAX support   Blacklist   Firewall   DDNS Support   Backup And Restore   Easy to install  Benefits:   Cost-effective Call Routing   Easy-to-navigate GUI   Simple Installation   Easy to integrate   Low power consumption for your green office  Applications:   Connect legacy PBX systems to VoIP services   Connect legacy PBX systems to remote sites over private VoIP links   Connect IP PBX systems to legacy TDM services   Phased transition from Legacy PBX to IP PBX  

APN Merchandising Limited

Featured here is a touch screen fixed PSTN and VoIP smart desktop telephone with Android operating system. It can be perfect integrated with mobile phones and offers an easy dialing experience. This smart desktop telephone has a large screen for making video calls and is excellent for conferencing use. 	 Touch screen fixed PSTN and VoIP smart desktop telephone with Android operating system. 	 USB connection for charging, also can connect with computers, support OTG. 	 Support WI-FI, WI-FI reset, WLAN connector. 	 5V USB charging output. 	 Ringer volume control, reset function. 	 Support headset calls and dial and hands-free calls and dial. 	 Hot spot sharing. 	 Perfect integration with mobile phones. 	 Easy dialing experience. 	 Unlimited intelligent application for making calls easily. 	 Directly call via contacts. 	 Contact information inquiry. 	 Easily set the black and white list, automatically identify and mark strange calls and refuse crank calls. 	 One key to import contacts. 	 Show caller ID. 	 Telephone message, call notes, telephone recording. 	 Historical call record. 	 Electronic photo album. 	 Video call. 	 Call recording. 	 Modern and functional design. 	 Great for video conferencing use. 	 Suitable for home and offices.  

Video Phone

Shenzhen Baoan Gaoke Electronics Co Ltd

Pictured is a VoIP phone with a hands-free dialing function. This VoIP phone will&nbsp;calculate conversation time automatically. The trendy design easily matches any modern office decor.   Supports&nbsp;DNS call waiting function.   Protocol support: SIP (RFD2543, ARFC3261).   Three-way conversation.  

NeoGate TG400 is a compact 4 channels VoIP GSM gateway that connects GSM network with VoIP. It is the ideal product for small and medium sized companies with heavy demands of calls to mobile networks. The cost-saving solution makes connection cheaper and efficient

Benefits
1) Cost savings - save phone bills tremendously with mobile-to-mobile calls and LCR
2) Back up - work as a cost-effective backup when the landline network goes down
3) Easy to install - everything can be easily set up in the Web based management interface
4) Easy to integrate - high compatibility with major IP PBX and soft switch brands

Specification   Number of GSM channels (max): 4   Network type: 850/900/1800/1900MHz   Protocol: SIP (RFC3261), IAX2   Transport protocol: UDP, TCP, TLS, SRTP   DTMF: RFC2833, SIP INFO, In-band   Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.   Echo cancellation: ITU-T G.168 LEC   LAN: 1 (10/100Mbps)   Network: static IP, DHCP client, firewall, VLAN, DDNS   QoS, open VPN   NAT traversal: static NAT, STUN   Size: 213x160x44mm   Power supply: AC 100-240V 50/60Hz 0.5A MAX   Operation range: 0&deg;to 40&deg;C, 32&deg; to 104&deg; F   Storage range: -20&deg; to 65&deg;C, -4&deg; to 149&deg; F   Humidity: 10-90% non-condensing  
Features   SIP server and SIP trunk supported   SIP peer mode supported   Calling type: VoIP to GSM, GSM to VoIP   GSM ports group manage, VoIP trunk group   Incoming /outgoing routing rules   SMS sending and receiving   Send bulk SMS   Gain adjustment   USSD   PIN modify   Carrier selection: auto/manual   Balance alarm   Caller ID/CLIR   Black list   Hotline   Call duration limitation   Call transfer   Call back   Call status display   Call detail record (CDR)   Call progress tone generation   Call duration limitation for SIM card/single call   LCR (least cost routing)   Top voice quality (EFR super sound)   SIP response code switch   Open API for SMS and USSD   Real open api protocol (based on asterisk)   IP blacklist   Network attack alert   System logs   Web based configuration  

VoIP-GSM Gateway

Suntra Int'l Development Ltd

This Internet telephone adaptor comes with a network interface set up by network for DHCP, PPPoE, or fixed IP network environment for connection to the network. It comes with auto configuration support (plug-n-play for no user setup when network access is available). Smart firmware management support is provided to&nbsp;eliminate the need for manual firmware upgrade.   Message prompts for ATA status (report IP address).   Compatibility in multi- vendor open architecture VoIP networks.   Dynamic DNS (DDNS) support for the use of domain name instead of IP address.   NTP and redundant DNS servers support .   QOS support (Diff-Serve, 802.1p/q VLAN).   NAT/ firewall traversal support with session border controller or stun server.   SIP standardization deployed throughout.   Advanced audio processing - jitter buffer with packet delay and packet loss concealment.   Voice packetization and flexible codec compression.   Dynamic and automatic negotiation of codec selection.   LAN Bridge &ndash; built in 2-port Ethernet hub.   FXS port for interface to a normal PSTN phone.   Bypass port for interface to a PSTN line (Optional).   PSTN caller ID signal generation for VoIP calls.   Calls from FXS port to both VoIP and PSTN networks.   Call transfer, call forward, call waiting for VoIP calls.   Easy local user customization of options - via web browser interface.   Open system SIP VoIP architecture for seamless integration into multi-vendor VoIP networks.   Standards include: SIP 2.0, TCP/UDP/IP, RTP/ RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, MD5 authentication.   Codec supported: G.711(a-law/u-law), G.729A/B, G.723.1, GSM FR   DTMF dialing in the format of RFC 2833, SIP Info, or In-band DTMF signals.  

Analog Telephone Adaptor

Fanvil Technology Co., Ltd.

 The IP phone is a sophisticated business VoIP communication terminal. The large dot matrix&nbsp;LCD display and soft-keys make&nbsp;it easy to use. This versatile VoIP phone is perfect for executives, office managers or administrative assistants who make and receive a large amount of calls. 
 
  2 headset interfaces (3.5mm jack or RJ9 connector). 
  6 memory keys. 
  Supports PoE. 
  Multi-language realizes localization. 
  SIP&nbsp;application: supports call forward/transfer/holding/waiting/conference, etc. 
  Flexible dial map and&nbsp;hotline. 
  Empty calling, black list and&nbsp;limit call. 
  No disturb function and caller ID display. 
  Incoming calls/outgoing calls/missing calls, each supports 100 records. 
  Supports phonebook of 500 records. 
  Supports conference and voice record on SIP server. 
 

IP Phone

One Source Telecom Limited

This white VoIP phone comes with basic functions such as call, redial and so on. The note feature records the important information effectively. This item is connected to the local network through the Ethernet.
 
  With 3 dialog keys. 
  Shows time, date and call information. 
 

Wired VoIP Phone

Neo Telecom Corporation

This wireless phone enables the high-quality VoIP service through a wireless-G network. It boasts unique features, such as 500 phone book memories and&nbsp;50 call histories. This functional cordless phone is perfect for busy families or offices.
 
  Call time display. 
  Dialed number display. 
  Redial function. 
  Call transfer, holding, waiting and forwarding functions. 
  SMS. 
  Language: English and Korean. 
  1.8&quot; color TFT LCD screen. 
  Resolution: 128 x 160. 
 

Wireless VoIP Phone

   The ALLO FXS/FXO to VOIP Gateway is standalone and fanless. Convert 4 analog ports (FXO or FXS module) to VOIP. Hardware DSP for LEC and transcoding.   Additional Features   Compatible with 3CX, Elastix, Asterisk, Trixbox, FreePBX..   On board 128ms Line Echo Cancellation DSP.   On board Transcoding (G711, G729) DSP.   4 analog ports (FXO or FXS).   T.38 fax relay.   Pre configured for almost plug and play experience.   Clean and easy to understand GUI with many options.   Caller ID.   DTMF Digit Detection and Generation.   SIP User Agent IETF RFC3261 compliant.  

FXS/FXO To VoIP Gateway - Analog Gateway 

The IP phone is a voice communication terminal. It comes with high voice quality. This versatile VoIP phone is perfect for executives, office managers or administrative assistants who make and receive a large amount of calls.
 
  Flexible dial map and&nbsp;hotline. 
  Empty calling reject and&nbsp;black list. 
  Limit call, no-disturb function and&nbsp;caller ID display. 
  Supports phonebook of 500 records. 
  Incoming calls/outgoing calls/missing calls, each supports 100 records. 
  Supports conference and voice record on SIP server. 
  Supports 10 kinds of ring type and 1 user ring. 
  Supports IAX2. 
  Supports message waiting indicator. 
  Supports SMS. 
  Supports speed dial. 
  Supports memo. 
  Multi-language realizes localization. 
 

The versatile IP phone features a wide LCD screen for clear display of information and call status. It comes with high voice quality. This versatile VoIP phone is perfect for executives, office managers or administrative assistants who make and receive a large amount of calls.
 
  Supports conference and voice record on SIP server. 
  Supports memory key, message waiting indicator and redial function. 
 

Featured here is an SIP phone that enables enterprises to take full advantage of the benefits of the VoIP telephony. This SIP phone&nbsp;has all capabilities that are necessary for daily business activities, such as call hold, call transfer, call swap, call list and 3-party conferencing functions.
 
  Telephone book with up to 100 entries. 
  Hand-free speaking, connection for microphone and headphone. 
  Function keys and programmable speed dials. 
 

SIP Phone

Shenzhen Voptel Techonology Co Ltd

VG5X VoIP Gateway is one product of VoIP VG series by VOPTECH. It is a carrier class VoIP gateway. Carriers and enterprises can use it to provide regular phone, fax, and IP Centrex services through broadband networks. VG5321 guarantees a smooth transition to NGN (Next Generation Network), therefore protects your long-term investments. When fully configured

VG5X is designed to meet the VoIP demands of telecom-munications carriers, value-added service providers, and mid- to -large -sized enterprises. Installed in a highly compact 1U chassis, and two power feeds (so end-users have the options to use one or two 70W power supply to protect system from power surge and to provide redundancy).

VG5X adopts embedded Linux operating system. Equipped with powerful hardware configuration and controlled by sophisticated software, the system supports multiple functions and provides solutions to a variety of application demands. In addition to supplementary functions like fashion ringing tones, Caller ID, Call Forwarding, and Call Waiting, it also supports features like, Routing, RADIUS Billing nterface, Broadcasting Second-stage


 Key Features 

Line Configuration Up to 96 ports, can be FXS + FXO mixed ports mode

Voice Processing Voice Codec: G.711, G.729A, G.723.1, GSM, iLBC
Echo Cancellation: G.168; Tail Length: 64 milliseconds
Dynamic Jitter buffer; Voice Activity Detector (VAD); and Comfort Noise Generator (CNG)

End user features Caller ID, Call Forwarding, Call Waiting, Hotline, Prefix Dialing, Do Not Disturb

Fax processing T.30, T.38 fax relay

Billing RADIUS Interface; Polarity reverse

Configuration methods Web based user Interface; Text based configuration

Remote Provisioning HTTP/WEB,Auto Provision, Remote Configuration, software upgrade, collects log files


 Singaling 

Off Hook Detection FXS Loop Start, DDI (Hong Kong)

VoIP Protocol SIPRFC3261), MGCP (RFC2705) 3GPP TS 24.228,TS 24.229

DTMF In-Band RFC2833,INFO (SIP)


 Other Special Feature 

Auto Dial DTMF mode

Dual Telephone Number FXS port can have two number for incoming calls ( PSTN and extension)


 Hardware 

Maximum User Line Length 3000 Meters

Ringer Equivalence Number 5 for short loop (300 meters), 3 for medium loop (1500 meters)

Ringing Voltage 60V RMS

Line interface RJ45, Per RJ45 provide 4 phone lines

Ethernet Interface Connector RJ-45 10/100 Base-T

Internal Memory 128 MB

Flash Memory 16 MB

Input voltage 110240V AC,5060Hz,Input current 1AMP Max

CPU Motorola PowerQuicc 8247

DSP TI 320VC5509AFeaturesNetwork

Power consumption Main unit 70W (Max),Expansion unit 55W (Max)

Dimension (H x W x D) 4.4&times;44&times;30 cm 1U height ,19 Chassis

Weight Net Weight7Kg,Gross weight 9Kg


 
 


VoIP Gateway

This wired VoIP phone is connected to the local network through the Ethernet. The self-labeling keys allow you to customize the telephone. The 2-line LED display shows the time, date and call information clearly.
 
  Durable and functional. 
  Cost effective. 
 

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