茂林(香港)线路板有限公司

   RJ45。   网络电缆及电话应用产品。  

双面电子线路板

进达电子制品有限公司

 	 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

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

&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
-&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

深圳市诚丰浩电子有限公司

10.1' Tablet Phone Android 10.0 PSTN+ 4G+SIP With handset (RJ9) 4G RAM+32G ROM (8G+64G optional) HD quality camera Support WiFi 2.4G /5G Desk / Wall mount Support GMS Support Google play store  

Smart Android Tablet Phone SIP Phone support 4G network

风青实业股份有限公司

This SIP phone combines the reliability of traditional telephones with the efficiency of internet-based technology, delivering a modern, flexible, and cost-effective solution for business communication. With minimal hardware and maintenance needs, it lowers long-distance and international call costs while enabling easy scaling by allowing businesses to add or remove users as needed. SIP Phone [Lower Communication Costs] Minimal hardware and maintenance requirements.[Scalable &amp; Efficient] Easily add or remove users as business needs evolve.[Wireless convenience] Perfect for spacious workplaces or settings where traditional wired connections are impractical.

SIP Phone

Eware Networks Limited

 	 Features Yealink HD Voice 	 Wall mountable  

Yealink SIP-T23P Enterprise HD IP Phone

Xenexx Corp Limited

 	 Allows two headsets to be plugged into the box Both of headsets can listen and talk 	 Two independent mute buttons for temporary call suspension 	 Has 8-position phone settings which are compatible with most desk phones 	 Has a 3.5mm jack for recording No battery required 	 Compatible with xenexx or other major brands headsets 	 Dimensions L 60mm W 62mm H 23mm  

Xenexx  Headset Training Box

 	 Features revolutionarily new design 	 Supports expansion modules 	 Wall mountable  

Yealink SIP-T48G Ultra-Elegant Gigabit IP Phone

 	 Features Full-HD 1080p30 video call Five-way audio/video mixed conferencing  

Yealink SIP VP-T49G Revolutionary Video Collaboration Phone

 	 Features revolutionarily new design 	 Yealink Optima HD voice integrated 	 Stand with 2 adjustable angles 	 Wall mountable  

Yealink SIP-T42G Ultra-Elegant Gigabit IP Phone

 	 Features Yealink Optima HD voice 	 Integrated stand with 2 adjustable angles 	 Wall mountable  

Yealink SIP-T27P Enterprise HD IP Phone

 	 Features Yealink Optima HD voice 	 Stand with 2 adjustable angles 	 Wall mountable  

Yealink SIP-T29G Enterprise HD IP Phone

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

Mercury Corporation

LTE+WiFi+VoIP+Ethernet Equipment (LTE to WiFi Gateway)

LTE CPE / LTE to WiFi Gateway

深圳市源创通信技术有限公司

The SinoV-S500 is an analog GSM cellular VoIP gateway that helps directly interconnect a SIP-supporting VoIP network with GSM networks. It features a strong housing to offer reliable protection to the internal components. Well engineered, this device ensures excellent performance.
 
  4/8 channels: 850/900/1,800/1,900MHz quad-band GSM cellular network. 
  SMS receiving/batch sending. 
  AT command compatible. 
  Support SIP registration and peer-to-peer. 
  Encryption: RC4, VOS, SinoV private, etc. 
  Hotline, phone book, speed dial, and dial plan. 
  Update firmware by local or Internet. 
  Field proven compatible with asterisk-based soft switches. 
 

GSM VoIP Gateway

深圳市沃宇科技有限公司

   VG1X-8 gateway is a perfect product for the telecom carriers to provide the residential telephony and fax services through the broadband networks such as FTTB, HFC, and ADSL. VG series gateway can be connected to the traditional PBXs to provide the voice solutions for enterprises, government, and schools. VG Series Gateway can also serve as a remote SIP terminal for IP-PBX solutions. VG Series Gateway is powerful hardware platform provides sufficient power for IP packetization, voice compression and echo cancellation even when the voice traffic is at the peak.   VG series SIP gateway is based on the embedded Linux operating system. With its powerful hardware platform and flexible software design VG Series Gateway can meet various voice application requirements.  Key feature:
   Line configuration 4 or 8 FXS analog ports; 4 FXO analog ports, mixed 4FXS/FXO analog port.   Voice processing voice codec: G.711, G.729A, G.723.1, GSM, iLBC.   End user features caller ID, call forwarding, call waiting, hotline, prefix dialing, DND.   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.  Signaling:
   Off hook detectio FXS loop start, DDI (Hong Kong).   VoIP protocol SIP (RFC3261), 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).   Support primary and backup the gateway can be configured and controlled by multiple soft switches.   Fashion ringing tones Can download and store 20 music files.  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 RJ11.   Ethernet interface connector RJ-45 10/100 Base-T.   Internal memory 32MB.   Flash memory 8MB.   Input voltage 110～240V AC, 50～60Hz, input current 1A max.   CPU Motorola Power Quicc 852.   DSP TI 5509A.   Power consumption: 5V/3A DC (A AC/DC convector is provided).   Dimension (H x W x D): 195 x 118 x 24 millimeters.   Weight: 0.65kg.  

8 FXS/FXO Port VoIP Gateway

Innotek Company Limited

The Solidtronic ST-RoIP3 Standalone Radio over IP Gateway is designed to interconnect mobile phone and 2-way radio users, or bridge 2-way radio systems across channels or sites.

The Solidtronic ST-RoIP3-Zello is equipped with the 3rd generation Super Dynamic Intelligent Digital Audio Tracking System(iDats III) PTT Control Technology, the Android 4.0.3 OS with 3rd party communication app Zello and powered by an ARM CPU, brings you the fast and stable push-to-talk communication between any device, anytime and anywhere over the world!

The key features of ST-RoIP3-Zello are:
@ Build-in ARM CPU standalone Radio-over-IP Network Gateway
@ Install in a second, no IT expert technique is required
@ Very easy operation
@ Very fast and stable PTT exchange performance by Super Dynamic iDats III PTT Control Technology
@ Embedded Android 4.0.3 OS
@ Pre-installed Zello app
@ Support different brands and different type radios

Zello Standalone RoIP Gateway

HUNG TAT INT'L (HK) LTD

This digital telephone answering machine features a metal keypad with a laser-etched backlight option.  The&nbsp;large-scale phone book stores up to 200 contact&nbsp;entries.  This telephone answering machine is a&nbsp;modern office assistant in&nbsp;busy companies.
 
  Handsfree speaker. 
  CID log:&nbsp;30 records. 
  Melody:&nbsp;4 monotones, 5 polytones, 9 optional polyphonic. 
  Intelligent low radiation. 
 

Digital Telephone Answering Machine

Shenzhen Cathay InfoTech Co., Ltd.

This phone is designed with a large monitor to provide touch-screen functions. The touch-screen design provides one-touch communication for patients and aged persons in hospitals or nursing homes. A video camera is set on top of the screen.
 
  Easy operation. 
  Available in various designs. 
 

TV Phone

YEASTAR TECHNOLOGY CO., LIMITED

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

   VG4X 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. VG4X guarantees a smooth transition to NGN (Next Generation Network), therefore protects your long-term investments. When fully configured.   VG4X 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).  Key features:
   Line configuration: up to 48 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.   End user features: generator (CNG).   Fax processing: caller ID, call forwarding, call waiting, hotline, prefix dialing, do not dist.   Billing: T.30, T.38 fax relay.   Configuration methods: radius interface; polarity reverse.   Web based user interface; text based configuration&quot;.   Remote provisioning: HTTP/web, auto provision, remote configuration, software upgrade, collects log files.  Signaling:
   Off hook detection: FXS loop start, DDI (Hong Kong).   VoIP protocol: SIP (RFC3261), 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).   Redundancy SIP server: support up to 10 SIP servers to enhance system reliability.   Fashion ringing tones: can download and store 20 music files.  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: 128MB.   Flash memory: 16MB.   Input voltage: 110～240V AC, 50～60Hz, input current: 1A max.   DSP: TI 320VC5509A.   CPU: Motorola PowerQuicc8247.   Power consumption: main unit 70W (max.), expansion unit 55W (max.).   Dimension (H x W x D): 4.4 x 44 x 30cm.   1U height, 19&rdquo; Chassis.   Weight: net weight: 9kg, gross weight: 14kg.   Environment requirement:   Operation environment: between 0～40℃ and humidity between 10%～95% (non condensing).   Non operation environment: 10～60℃.  Package:
   Unit weight (kg): 4.04.   L: 53.   W: 37.   H: 15.   Unit volume weight (kg): 5.9.  

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