Enlighten Company Limited

Model: EL-WCP-IPInput power: PoE(802.3af)Protocol: TCP-IPVoIP protocol: SIPEnvironment Noise CancellationEcho CancellationIP-65 water-proofRed re-assurance visible LEDBig alarm button with Reset

IP based Call Point with VoIP

Microtech Technology Co Ltd

MT-3399 is a high-end development board based on Rockchip RK3399. MT-3399 is the first 64-bit development board with super high performance based on industrial control. MT-3399 is incorporated with six cores: four A53 cores and two A72 cores. Its dominant frequency is up to 2GHz.
RK3399 is rated monster level in terms of CPU and GPU. Incorporating two Cortex-A72 large cores and four Cortex-A53 small cores, the CPU has been greatly improved in terms of integers, floating decimals, and memory as well as overall performance, power consumption, and core area.
The GPU is incorporated with a new-generation high-end ARM image processor Mali-T860, which contains four cores, integrated with more bandwidth compression technologies, such as intelligent superposition, ASTC, and local pixel storage. The GPU also supports more graphic and computing interfaces and its overall performance is 45% greater than that of the last generation products.
Although these capabilities are called top capabilities of the chip solutions of the same kind, they are not the key features of RK3399. RK3399 is highlighted in the Type-C interface, embedded PCI-E interface, and two cameras that support gesture recognition, which help overturn the experience in using the game box products. RK3399 also has other new features, such as supporting LPDDR4 memory, ahead of the current mainstream products.
Built upon high configuration and the improvement in the overall performance as well as comprehensive layout, RK3399 is born to be an all-rounder. Apart from the panel computer, VR, TVBOX, and laptop, RK3399 is applied to a diversity of terminals in the industry and consumption fields, including intelligent household appliances, advertising machines/integrated machines, financial POS terminals, vehicle-mounted control terminals, thin clients, VOIP video conferencing, security/surveillance/police affairs and Internet of Things.
 Android System Configuration  	 CPU: RK3399 	 Clock speed:&nbsp; Four A53 (1.4GHz) cores + two A72 (2GHz) cores 	 Memory:&nbsp; 2GB for standard; 4GB be customized 	 Storage: 4GB/8GB/16GB eMMC optional, 16GB for the standard configuration 	 Power supply IC: Using RT808 and supporting dynamic frequency modulation  

OEM/ODM Customized Motherboard RK3399 Android System 2GB DDR 16GB eMMC Super Development Board

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

Fanvil Technology Co., Ltd.

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. 
 

IP 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.  

VoIP 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

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

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

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 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. 
 

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

   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 

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

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

Nanjing Hanlong Technology Co., ltd

 Displayed is an IP phone that comes with a user-friendly graphic LCD screen. With two ethernet ports of different speeds, this telephone can easily help you make internet calls of satisfactory quality.&nbsp;This wall-mounted VoIP phone also has an HD voice system so that you can hear better.&nbsp; 
 
  HD IP phone with 128 x 96 graphic LCD screen. 
  Dual ports: 10Mbps and 100Mbps ethernet ports. 
  3 VoIP accounts. 
  Multiple languages. 
  Wall mounted. 
  14 programmable keys. 
  Support BLF and BLA. 
  HD voice: HD handset, HD speaker, HD codec and HD wideband codec (G.722). 
  Call waiting, call transfer, call forward, voicemail, intercom, SMS LDAP or Xml phonebook. 
  Open VPN and&nbsp;TLS-SRTP. 
  Auto-provision via TFTP, FTP, HTTP, HTTPS and PNP. 
 

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. 
 

This wired VoIP phone is equipped with self-labeling keys for maximum flexibility. The 2-line display shows the time, date and call information clearly. This item is connected to the local network through the Ethernet.
 
  With 3 dialog keys. 
  Includes an integrated mini switch for connecting to computer. 
 

 Displayed is an IP phone that comes with a large LCD color screen for your convenience. Featuring dual ports and ethernet ports, this phone supports up to four VoIP accounts and has HK voice functions, including an HD handset and speaker. This high quality IP phone is also available in multiple languages. 
 
  Gigabit IP Phone. 
  3.5&rdquo;TFT LCD&nbsp;of&nbsp;480 x 320 pixels and&nbsp;262K colors. 
  Dual ports. 
  Gigabit ethernet ports. 
  4 VoIP accounts. 
  Multiple languages. 
  14 programmable keys. 
  Support BLF and BLA. 
  HD voice: HD handset, HD speaker, HD codec&nbsp;and HD wideband codec (G.722). 
  Call waiting, call transfer, call forward, voicemail, intercom, SMS LDAP and Xml phonebook. 
  Open VPN and&nbsp;TLS-SRTP. 
  Auto-provision via TFTP, FTP, HTTP, HTTPS and PNP. 
 

   The revolutionary FXS/FXO to VOIP card will be detected as a network card in all Windows and Linux systems. The OS will install the driver (network card) then you can use it with your favorite opensource PBX as a regular VOIP channel. The card itself is configured with a script. Hardware DSP for LEC (carrier grade 128ms) and transcoding.  Additional Features: 
   Compatible with 3CX, Elastix, Asterisk, Trixbox, FreePBX.   Windows/Linux compatible (all versions).   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 Card - Analog Card

  SIP standardization deployed throughout  Sustain VoIP calls in noisy unmanaged networks, supporting up to 40% packet loss and up to 500ms of latency  Supports TOS bit for improved QoS in managed networks (Differentiated services of "DiffServ")  Voice packetization & flexible compression  Dynamic & automatic negotiation of codec selection  LAN bridge   built in 2 port ethernet hub   (Optional)  Caller ID receive and display  Alphanumeric LCD  Hands free dialing  Phone book memory dialing  Incoming, outgoing and missed call logs  

VoIP SIP Phone

 Displayed is an IP phone that comes with a user-friendly color screen. Supporting up to three VoIP accounts, this phone has 14 programmable keys and an inclusive HK voice system, including a&nbsp;handset, speaker, codec and wideband codec. With a fast ethernet port, this IP phone is&nbsp;a nice choice for those living far away from one another. 
 
  Enterprise color IP phone. 
  3.5&rdquo; TFT LCD&nbsp;with&nbsp;480 x 320 pixels and&nbsp;262K colors. 
  Dual ports. 
  10/100Mbps ethernet port. 
  3 VoIP accounts. 
  Multiple languages. 
  Wall-mounted. 
  14 programmable keys. 
  Support BLF and BLA. 
  HD voice: HD handset, HD speaker, HD codec and&nbsp;HD wideband codec (G.722). 
  Call waiting, call transfer, call forward, voicemail, intercom, SMS LDAP or Xml phonebook. 
  Open VPN and&nbsp;TLS-SRTP. 
  Auto-provision via TFTP, FTP, HTTP, HTTPS and PNP. 
 

   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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