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Electronics Suppliers and Manufacturers

Suntra Int'l Development Ltd


   This is a business class (SIP) IP telephone.  The phone features handsfree dialing.  Alphanumeric LCD  Lan bridge build in 2 port ethernet hub  Phonebook memory dialing  SIP sandardization deployed throughout  
 

Business Class (SIP) IP Telephone


   This is a business class (SIP) IP telephone.  It is equipped with headset jack.  Multiple lines (up to 4 lines)   Register to a single server or multiple service mode   Extra large LCD display (128*64 dot,v/a 74*40)  SIP sandardization deployed throughout  PO (no adaptor required)  
 

Suzhou International Science Park

This outlined IP video phone is designed with advanced features. The large LCD touch screen and keypads ensures enhanced user friendliness. The phonebook can save up to 2,000 records. This IP video phone is ideal for office use.
 
  Telephone features: forward and transfer calls, conference call, paging and intercom, etc. 
  Blacklist, whitelist, DND. 
  Call history: incoming, outgoing and missed (each 1,000 entries). 
  Import and export phonebook. 
  External phone book: XML. 
  Photo frame. 
 

IP Video Phone

Wahkow Science & Technologies Limited

This is an IP phone with an elegant and practical design. It features a black housing with a sophisticated control panel and a 3.5-inch LCD display. Well engineered, this telephone delivers excellent performance with high stability.
 
  Gigabit color. 
  3.5&rdquo; TFT-LCD screen. 
  480 x 320 pixels. 
  262K color dual-port gigabit Ethernet ports. 
  4 VoIP accounts. 
  Multi-language 14 programmable keys. 
  Support BLF/BLA HD voice: HD handset, HD speaker, and HD codec. 
  HD wideband codec: G.722. 
  Call waiting, call transfer, call forward, voicemail, intercom, and SMS LDAP/Xml phonebook. 
  Open VPN. 
  TLS-SRTP Auto-provision via TFTP/FTP/HTTP/HTTPS/PNP. 
 

IP Phone

Shenzhen Voptel Techonology Co Ltd

VI2006 is ideal for both the small office and general enterprise users.Enterprise SIP phone's new design and enhanced features address the need for an elegant IP handset solution for the executive office at a highly competitive price.


VI2006 is an Internet based voice network phone terminal, which adopts multiple voice control protocols and voice compression coding methods to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.


VI2006 supports IAX2&amp;SIP protocols,support Bridge and Router model.Support IAX2 and dual public server, offers Two Ethernet interfaces and is compatible with various softswitch systems and VoIP voice gateways to provide broadband IP voice service.


VI2006 supports traditional telephony features such as Flexible dial map, Hotline, Empty calling reject, Black list for reject authenticated call, limit call, No disturb, Caller ID as well as advanced enterprise features such as 2 independent SIP accounts, 3-way and conferencing. 

Specifications

Support SIP 2.0 (RFC3261) and correlative RFCs
Codec:G.711 A/U Law, G.723.1, G.729a/b,G.722,G.722.1, G.726
Echo cancellation: Support G.168, and Hands-free can support 96ms, Hand free Speaker Phone
Support Voice Gain Setting, VAD, CNG
Full duplex hands-free speakerphone
NAT transverse:support STUN client
SIP support SIP domain, SIP authentication (none,basic, MD5), Peer to Peer/ IP call
SIP support 2 SIP lines. Can connect to SIP1 and SIP2server at the same time
DTMF:Support SIP info, DTMF Relay, RFC2833
SIP application: support Call forward/ transfer/holding/ waiting / 3 way talking
Call control features: Flexible dial map, Hotline,Empty calling reject, Black list for reject authenticatedcall, limit call, No disturb, Caller ID
Support Phonebook 500 recordsIncoming calls / Outgoing calls /Missing calls. 
Each support 100 records
Support 10 kind of ring type and 1 user ring
Support SRTPSupport SMS
Support Speed Dial

Network Features

Support basic NAT and NAPT
Support PPPoE for xDSL
Support DHCP Client on WAN
Support DHCP Server on LAN
Support VLAN (optional: voice vlan/data vlan) QoS with Diff Serv
Support DMZSupport main DNS and secondary DNS server
Support DNS Relay, SNTP Client, Firewall
Network tools in telnet server: Including ping, trace route,telnet client

Maintenance&amp;Management

Web ,telnet and keypad management
Management with different account right
Upgrade firmware through HTTP, FTP or TFTP. Telnet 
Remote management/Upload/download setting file
Safe mode provide reliability
Supoort Auto Provisioning (upgrade firmware orconfiguration file)
Support Syslog

 

Allodotcom Inc.

  - As shown is a telephone adapter.  - The telephone adapter is equipped with a 2x ethernet LAN interface - RJ45.  - Out-of-band DTMF (RFC 2833)   - 2 FXS ports & 1 FXO port  - Adaptive jitter buffer  - SIP Phone Number per FXS port  - User-friendly web panel  - DTMF out-of-band (RFC 2833)  - DHCP server for LAN  - Operating System: Linux 2.4.26  - IP Address Reservation by MAC Address  - Simultaneous active calls on FXS1 and FXS2  - PSTN pass-through  - Firmware upgrade via RFC 1867 upload  - SIP compatible  

VoIP Analog Telephone Adapter

CEM Solutions Pvt Ltd.

IP telephony applications commonly require the use of multiple voice codecs, used to digitally compress voice signals and save on bandwidth. Voice signals from the PSTN come in the form of the G.711 codec, but the VoIP terminal equipment and networks can support a variety of different voice codecs, such as G.729. The VoIP infrastructure needs the capability to mediate between endpoints supporting different codecs. 

Our Transcoding Cards are designed to handle complex codec translation, using dedicated DSP resources, which would otherwise be processed by host CPU in software. This card greatly reduces the MIPS or CPU consumption, so that it can be used for handling other tasks. 

Our Transcoding Cards are available in PCI and PCI Express form factors, converts simultaneous channels of transcoding from one type of codec (e.g. G.711) to another (e.g. G.729), without affecting latency or using up precious host CPU resources. 

 Key Features  
   This card can handle 64, 128, 256 or 400 Trascoding Sessions   Supported codecs: G711u/a G.723 G.726 G.722 G729 AB   PCI 2.2/PCIe compliant   Integrates with Open Source platforms such as Asterisk, Freeswitch, Elastix , YATE and more  

Transcoding Card for Asterisk PCI/PCIe

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. 
 

HK Retail Technology Industry Association Ltd/GS1 HK

The wafer UC manager provides call traffic statistics, call billing and QoS analysis functions. It also provides a management platform of the enterprise IP telephone network. The UC manager offers analysis solutions of investment evaluations for the enterprises based on the voice communication network.
 
  High quality. 
  Professional and reliable. 
 

Wafer UC Manager

The wafer VoizMate is a daily Cisco IP phone application platform. It combines with the Cisco IP telephony solutions to make full use of IP phone communications for enterprises including more value-added services. It can satisfy more requirements of corporation for service integration and information bulletin, finally improving the corporate efficiency and competitiveness.
 
  Combines with the Cisco IP telephony solutions to make full use of IP phone communications for enterprises. 
  High quality. 
 

Wafer VoizMate

Vogtec (H.K.) Co., Limited

 	 Deeply customized Android system with best compatibility, stability, and extensibility 	 2.8&quot; 320 x 240LCD with touch panel options 	 TDD LTE / FDD / LTE, CDMA2000, WCDMA fix line 	 Inbuilt mobile antenna, ext. WiFi&nbsp;antenna 	 Support both voice and data via WiFi hots-pot or WAN port (optional) 	 Specialized for Tel co fix line with network lock / cell lock / pin lock 	 Support both Bluetooth and RJ9 headset 	 VoIP over LTE / WCDMA / CDMA2000 data network 	 FoTA 	 Wire Type: Cordless 	 Voice Mail Function: Yes  

VoIP Over LTE and WiFi 2.4 / 5G Desktop Phone 

VoIP Desktop T12 	 Generous 128x64 pixel back light graphical LCD display or 2.8&#39; TFT color LCD 	 Multi Languages 	 SIP (RFC3261),TCP/IP/UDP, RTP, HTTP, ARP, ICMP, DNS (A record and SRV),DHCP(both client and server), PPPoE, TFTP, NTP 	 NAT traversal (STUN, etc), server fail-over, SIP presence (SIMPLE), and more 	 2 lines indicators with individual SIP account profiles 	 Soft keys, navigation keys and Line Keys can defined by user 	 Call waiting, hold, mute, transfer (blind or attended), forward, 3-way conferencing and more 	 Full duplex speaker phone w/advanced audio echo cancellation 	 Headset jack RJ9 for broad headset compatibility 	 Large phonebook (up to 500 contacts) and call history (up to 100 records) 	 Automated provisioning using TR-069 or encrypted XML configuration file, SRTP and TLS for advanced security protection, 802.1x for media access control 	 And many more enterprise grade features  

VoIP Desktop

   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

   VG3X 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 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 features:
   Line configuration: 2 FXS analog ports.   2 FXO/2FXS analog ports, 1FXS/1FXO analog port Line Configuration.   Voice processing: voice codec: G.711ALaw, G.711Ulaw, G.729 (Note1) and G.729B (Note1).   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, DND.   Billing: T.30, T.38 fax relay.   Configuration methods: radius interface; polarity reverse.   Web based user interface; text based configuration.   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: 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: 2 RJ-45 10/100 base-T port.   Internal memory: 32MB.   Flash memory: 4MB.   Input voltage: 110～240V AC, 50～60Hz, input current 1A max.   CPU: ATMEL AT91SAM9G20B with 400MHz.   Power consumption: 5V/3A DC (A AC/DC convector is provided).   Dimension (H x W x D): 150 x 109 x 30 millimeters.   Weight: 0.3kg.   Environment requirement:   Operation environment: between 0～40℃ and humidity between 10%～95% (non condensing).   Non operation environment: 10～60℃.   Weight and size:   Unit weight (kg): 0.44.   L: 24.   W: 17.   H: 6.   Unit volume weight (kg): 0.5.  

VoIP Voice Gateway with 2 FXS/FXO Port

The pictured IP video phone is fitted with the professional Broadcom VoIP and video chipsets chipsets. It is equipped with a 7&quot; digital 800 x 480 TFT LCD touch screen for brilliant performance. The IP video phone is an ideal choice for business meeting use.
 
  Android OS. 
  SIP supports 5 SIP servers. 
  VoIP/PSTN dual modules supported. 
  Visual angle adjustable display screen. 
  Rotatable CMOS sensor camera with 2 mega pixels. 
  Video resolution: QCIF, CIF, 4CIF, DCIF. 
  Image format: JPEG, PNG, GIF, BMP. 
  Support phonebook 2,000 records. 
 

This is an IP phone with a durable structure. It features a black housing with user-friendly buttons and a high-quality&nbsp;display. This well-made telephone promises an extended service life.
 
  Color HD phone. 
  3.5&quot; TFT-LCD screen. 
  480 x 320 pixels. 
  262K colors. 
  dual-port 10/100Mbps Ethernet port. 
  4 VoIP accounts. 
  Multiple languages. 
  Wall-mounted. 
  4 programmable keys. 
  Support BLF/BLA HD voice: HD handset, HD speaker, and HD codec. 
  HD wideband codec: G.722. 
  Call waiting, call transfer, call forward, voicemail, intercom, and SMS LDAP/Xml phonebook. 
  Open VPN. 
  TLS-SRTP auto-provision via FTP/FTP/HTTP/HTTPS/PNP. 
 

Fanvil Technology Co., Ltd.

As shown here is an IP video phone that runs on the Android 2.1 operating system. It is embedded with a rotatable 2 mega-pixel CMOS sensor camera for added performance. This IP video phone features a 7&rdquo; 800 x 480 TFT LCD touch screen, ensuring vivid image presentation.
 
  SIP supports 5 SIP servers. 
  Visual angle adjustable display screen. 
  Video resolution: QCIF, CIF, 4CIF, DCIF. 
  Video codec: H.263, H.264, VC-1, MPEG4. 
  Image format: JPEG, PNG, GIF, BMP. 
  Phonebook supports 2,000 records. 
 

Displayed here is an IP video phone that is equipped with a rotatable CMOS sensor camera with 2 mega pixels. The 7&quot; digital TFT 800 x 480 pixels LCD touch panel allows brilliant image display. This IP video phone operates on the Android 2.1 operating system.
 
  Supports SD card slot. 
  Video resolutions: GCIF, CIF, 4CIF, DCIF. 
  Video codec: H.263, H.264, VC-1, MPEG4. 
  Image format: JPEG, PNG, BMP. 
  Phonebook supports 2,000 records. 
 

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

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  

Kyung Jin Electron Co., Ltd.

 Experience wireless freedom with KJ's latest Bluetooth headset, KB-10. It has great mobility and sound suitable to work at both offices and home.    

KB-10

Outlined here is an IP phone with a simple yet practical design. The large LCD display screen allows easy viewing. This IP phone is easy to install and operate.
 
  Support 2.0 (RFC32961) and correlative RFCs. 
  Support IA x2.Support 4 sip lines. 
 

Kingtel Corporation

   The phone uses SIP protocol.   The IP phone complies with all well-known IP-PBX brands.   The phone supports auto-provision.   The phone is rated as high-quality speakerphone as commercial grade.   The phone supports POE-power over ethernet.   Single line speaker phone.  

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