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電子產品供應商和製造商

Electronics

電訊服務

搜羅全球頂尖電訊服務公司提供的電訊服務資訊。您可在貿發網採購平台即時諮詢可靠和專業的產品供應商，以獲取詳盡的資料及報價。

電訊服務供應商和製造商

Telecommunication

電話

elec

搜羅全球頂尖電話公司提供的電話資訊。您可在貿發網採購平台即時諮詢可靠和專業的產品供應商，以獲取詳盡的資料及報價。

電話供應商和製造商

Phones

進達電子製品有限公司

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

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

風青實業股份有限公司

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

Fanvil Technology Co., Ltd.

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. 
 

IP Phone

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

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 IP phone can register up to 4 SIP accounts. It features a large LCD screen and functional keys for added performance. This IP phone is convenient to use.
 
  SIP 2.0 (RFC3261). 
  HD voice on all lines. 
  Echo cancellation: G.168 compatible with the LEC, including hands-free (AEC). 
  VGS, VAD, CNG. 
  Speakerphone full-duplex. 
  Phonebook up to 500 entries. 
  Customized logos are welcome. 
 

Huizhou City Huiyang District Meisiqi Industry Development Co., Ltd.

MSQ VoIP phone series are feature-rich, cost-effective SIP phones specially designed for ITSPs, small to medium businesses and residential application. It supports 5 user accounts, auto-provision as well as zero configuration and easy maintenance. It offers high quality audio, a broad range of voice codes and phone features to general level users. 
   Two RJ45 ports with router built-in, one for Internet, one for PC.   One RJ11 port FXO port.   POE (power over Ethernet).  Network protocol: 
   SIP V1(RFC2543), V2(RFC3261).   IP/TCP/UDP/RTP/RTCP.   IP/ICMP/ARP/RARP/SNTP.   TFTP client/DHCP client/PPPoE client.   Telnet/HTTP server.   DNS client.   NAT.  SIP server:
   Registrar server: 5 SIP user accounts.   Outbound proxy mode.   Power over Ethernet (optional).  Code:
   G.711.   G.723.1/iLBC.   G.726.   G.729A.   G.729B adds VAD, CNG to G.729.  Configuration: 
   Web browser.   Console/telnet.   IVR/keypad.   Auto-provision.  IP assignment: 
   Static IP.   DHCP.   PPPoE.  NAT traversal: 
   STUN.   QoS.   ToS field.  Dial method: 
   Direct IP call without SIP proxy.   Dial number via SIP server.   Dial URL via SIP server.  DTMF function: 
   In-band DTMF.   Out-of-band DTMF.   SIP info.  Phone features: 
   Caller ID.   Call hold.   Call waiting.   Call forward.   Speaker phone.   3-way conference.   Redial and flash.   Speed dial.   Phone book.   Volume adjustment.   Ring tone selection.  Voice quality: 
   VAD.   CNG.   LEC.   Packet loss compensation.   Adaptive jitter buffer.  Firmware upgrade: 
   TFTP.   Console.   HTTP.   Auto-provision.  Security: 
   HTTP 1.1 basic/digest authentication for web setup.   MD5 for SIP authentication (RFC 2069/RFC 2617).  Interface: 
   Power input: 100-240V AC, output power: DC-5.5V, 1A.   WAN RJ45, 10/100M port.   LAN/PC RJ45, 10/100M port.   TEL IN (FXO) RJ11 port.  Certificate:
   CE.   FCC.  

VoIP SIP Phone POE with PSTN 

深圳市沃宇科技有限公司

Asterisk, Elastix, 3CX compatible.
VoIP phone with 3 SIP lines &amp; 1 IAX2 line.
Support VLAN, VPN (L2TP), POE.
RJ9 headset.
2RJ 45 ports.
Large dot matrix LCD display.
 
Basic features: 
   Support SIP 2.0 (RFC3261) and correlative RFCs.   Support IA x 2.   SIP supports 3 SIP servers. Can connect to 3 SIP servers at the same time.   Codec: G.711A/u, G.723, G.729a/b, G.722.   Echo cancellation: G.168 compliance in LEC, additional acoustic echo cancellation.   Support voice gain setting, VAD, CNG.   Full duplex hands-free speaker phone.   SIP support SIP domain, SIP authentication (none, basic, MD5).   DTMF mode: support DTMF_RELAY, SIP_INFO, RFC_2833.   Network features: 
   WAN/LAN: 10/100M Ethernet ports, supports Route and Switch mode.   Support PPPoE for xDSL.   Support basic NAT and NAPT.   NAT transverse: support STUN client.   Support DHCP client on WAN.   Support DHCP server on LAN.   Support main DNS and secondary DNS server.   Support VLAN.   Support DMZ.   Support DNS relay, SNTP client, firewall.   Support VPN (L2TP).   QoS with DiffServ.   Network tools in telnet server: including ping, trace route, telnet client.   Advanced application features: 
   Large dot matrix LCD display and softkeys make user easier to use.   2 headset interface (3.5mm Jack or RJ9 connector).   6 memory keys.   Support POE.   Multi language realizes localization.   SIP application: support call forward/transfer/holding/waiting/conference/.   Call control features: flexible dial map, hotline, empty calling no. reject service.   Incoming calls/outgoing calls/missing calls each support 100 records.   Support phone book 500 records.   Support conference and voice record on SIP server.   Support SMS.   Maintenance and management.   Web, telnet and keypad management.   Management with different account right.   Upgrade firmware through POST mode.   Telnet remote management/upload/download setting file.   Safe mode provide reliability.   Support auto provisioning (upgrade firmware through HTTP, FTP or TFTP).   Support syslog.  

VoIP Phone Compatible with Asterisk

   VI2007 is SIP phone 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.   VI2007 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.   VI2007 supports IAX2 &amp; SIP protocols, support bridge and router model. Support IA x 2 and dual public server, offers two Ethernet interfaces and is compatible with various soft switch systems and VoIP voice gateways to provide broadband IP voice service.   VI2007 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 96m, hand free speaker phone.   Support voice gain setting, VAD, CNG.   Full duplex hands-free speaker phone.   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 SIP2 server 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 authenticated call, limit call, no disturb, caller ID.   Support phonebook 500 records incoming calls/outgoing calls/missing calls.   Each support 100 records.   Support 10 kind of ring type and 1 user ring.   Support SRTP.   Support 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 DiffServ.   Support DMZ.   Support 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.   Support auto provisioning (upgrade firmware or configuration file).   Support syslog.  

Eware Networks Limited

The SIP-T46G is Yealink latest revolutionary IP Phone for executive users and busy professionals. New designs appears the commerce, with highresolution TFT color display, delivering a rich visual experience. Yealink Optima HD technology enables rich, clear, life-like voice communications. Supports Gigabit Ethernet, a variety of device connections, including EHS headset and USB. With programmable keys, the IP Phone supports vast productivity-enhancing features.


 Features  	 Revolutionarily new design 	 Yealink Optima HD voice 	 Dual-port Gigabit Ethernet 	 4.3&quot; 480 x 272-pixel color display with backlight 	 Built-in a USB port, support Bluetooth headset (Through USB Dongle) 	 Up to 16 SIP accounts 	 Paper label free design 	 PoE(802.3af) class support 	 Headset, EHS support 	 Supports expansion modules 	 Stand with 2 adjustable angles 	 Wall mountable 	 Simple, flexible and secure provisioning options  
 Specification 

Audio Features 	 HD voice: HD handset, HD speaker 	 Codecs: G.722, G.711(A/&mu;), G.723, G.729AB, G.726, iLBC 	 DTMF: In-band, Out-of-band(RFC 2833) and SIP INFO 	 Full-duplex hands-free speakerphone with AEC 	 VAD, CNG, AEC, PLC, AJB, AGC  
Phone Features 	 16 VoIP accounts 	 Call hold, mute, DND 	 One-touch speed dial, hotline 	 Call forward, call waiting, call transfer 	 Group listening, SMS, emergency call 	 Redial, call return, auto answer 	 3-way conferencing 	 Direct IP call without SIP proxy 	 Ring tone selection/import/delete 	 Set date time manually or automatically 	 Dial plan 	 XML Browser 	 Action URL/URI 	 RTCP-XR (RFC3611), VQ-RTCPXR (RFC6035)  

Yealink SIP-T46G Ultra-Elegant Gigabit IP Phone

   VI2006 is IP phone ideal for service provider and small offices. 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 IA x 2 &amp; SIP protocols, support bridge and router model. Support IA x 2 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 speaker phone.   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 SIP2 server 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 authenticated call, limit call, no disturb, caller ID.   Support phonebook 500 records incoming calls/outgoing calls/missing calls.   Each support 100 records.   Support 10 kind of ring type and 1 user ring.   Support SRTP.   Support 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 DiffServ.   Support DMZ.   Support 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.   Support auto provisioning (upgrade firmware or configuration file).   Support syslog.  

Low Cost VoIP Phone in Promotio

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

 

   MSQ VoIP phone series are feature-rich, cost-effective SIP phones specially designed for ITSPs, small to medium businesses and residential application.   It supports 5 user accounts, auto-provision as well as zero configuration and easy maintenance.   It offers high quality audio, a broad range of voice codes and phone features to general level users.   Due toits excellent cost-performance ratio and high performance, it is absolutely a good choice for you.   Two RJ45 ports with router built-in, one for internet, one for PC.   POE (power over Ethernet) optional.  Network protocol:
   SIP v1 (RFC2543), v2 (RFC3261).   IP/TCP/UDP/RTP/RTCP.   IP/ICMP/ARP/RARP/SNTP.   TFTP client/DHCP client/PPPoE client.   Telnet/HTTP server.   DNS client.   NAT.  SIP server:
   Registrar server: 5 SIP user accounts.   Outbound proxy mode.   Power over Ethernet (optional).  Code:
   G.711.   G.723.1/iLBC.   G.726.   G.729A.   G.729B adds VAD, CNG to G.729.  Configuration:
   Web browser.   Console/telnet.   IVR/keypad.   Auto provision.  IP assignment:
   Static IP.   DHCP.   PPPoE.  NAT traversal:
   STUN.   QoS.   ToS field.  Dial method:
   Direct IP call without SIP proxy.   Dial number via SIP server.   Dial URL via SIP server.  DTMF function:
   In-band DTMF.   Out-of-band DTMF.   SIP info.  Phone features:
   Caller ID.   Call hold.   Call waiting.   Call forward.   Speaker phone.   3 way conference.   Redial and flash.   Speed dial.   Phone book.   Volume adjustment..   Ring tone selection.  Voice quality:
   VAD.   CNG.   LEC.   Packet loss compensation.   Adaptive jitter buffer.   Firmware upgrade.   TFTP: console.   HTTP: auto-provision.  Security:
   HTTP 1.1 basic/digest authentication for web setup.   MD5 for SIP authentication (RFC 2069/RFC 2617).  Interface:
   Power input: 100-240V AC, output power: DC 5.5V, 1A.   WAN RJ45,10/100M port.   LAN/PC RJ45,10/100M port.   Certificate: CE, FCC  

IP Phone with POE and PSTN Port

深圳市寶安高科電子有限公司

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

Allodotcom Inc.

The IP phone&nbsp;displays incoming and missed calls with date and time. The intuitive control panel ensures easy operation. This IP phone is a great choice for your home and office.

 
  Call log. 
  Hold function. 
  Hot keys. 
 

MSQ VoIP gateway series are feature-rich, cost-effective SIP gatway specially designed for ITSPs, small to medium businesses and residential application. It supports 5 user accounts, auto-provision as well as zero configuration and easy maintenance. It offers high quality audio, a broad range of voice codes and phone features to general level users. 
   Two RJ45 ports with router built-in, one for Internet, one for PC.   2 FXS + 1FXO port.  Network protocol: 
   SIP V1(RFC2543), V2(RFC3261).   IP/TCP/UDP/RTP/RTCP.   IP/ICMP/ARP/RARP/SNTP.   TFTP client/DHCP client/PPPoE client.   Telnet/HTTP server.   DNS client.   NAT.  SIP server:
   Registrar server: 5 SIP user accounts.   Outbound proxy mode.  Code:
   G.711.   G.723.1/iLBC.   G.726.   G.729A.   G.729B adds VAD, CNG to G.729.  Configuration: 
   Web browser.   Console/telnet.   IVR/keypad.   Auto-provision.  IP assignment: 
   Static IP.   DHCP.   PPPoE.  NAT traversal: 
   STUN.   QoS.   ToS field.  Dial method: 
   Direct IP call without SIP proxy.   Dial number via SIP server.   Dial URL via SIP server.  DTMF function: 
   In-band DTMF.   Out-of-band DTMF.   SIP info.  Voice quality: 
   VAD.   CNG.   LEC.   Packet loss compensation.   Adaptive jitter buffer.  Firmware upgrade: 
   TFTP.   Console.   HTTP.   Auto-provision.  Security: 
   HTTP 1.1 basic/digest authentication for web setup.   MD5 for SIP authentication (RFC 2069/RFC 2617).  Interface: 
   Power input: 100-240V AC, output power: DC-12V, 1A.   WAN RJ45, 10/100M port.   LAN/PC RJ45, 10/100M port.  Certificate:
   CE.   FCC.  

VoIP Gateway SIP IP ATA 

MSQ VoIP gateway series are feature-rich, cost-effective SIP gatway specially designed for ITSPs, small to medium businesses and residential application. It supports 5 user accounts, auto-provision as well as zero configuration and easy maintenance. It offers high quality audio, a broad range of voice codes and phone features to general level users. 
   Two RJ45 ports with router built-in, one for Internet, one for PC.   1FXS Port  Network protocol: 
   SIP V1(RFC2543), V2(RFC3261).   IP/TCP/UDP/RTP/RTCP.   IP/ICMP/ARP/RARP/SNTP.   TFTP client/DHCP client/PPPoE client.   Telnet/HTTP server.   DNS client.   NAT.  SIP server:
   Registrar server: 5 SIP user accounts.   Outbound proxy mode.  Code:
   G.711.   G.723.1/iLBC.   G.726.   G.729A.   G.729B adds VAD, CNG to G.729.  Configuration: 
   Web browser.   Console/telnet.   IVR/keypad.   Auto-provision.  IP assignment: 
   Static IP.   DHCP.   PPPoE.  NAT traversal: 
   STUN.   QoS.   ToS field.  Dial method: 
   Direct IP call without SIP proxy.   Dial number via SIP server.   Dial URL via SIP server.  DTMF function: 
   In-band DTMF.   Out-of-band DTMF.   SIP info.  Voice quality: 
   VAD.   CNG.   LEC.   Packet loss compensation.   Adaptive jitter buffer.  Firmware upgrade: 
   TFTP.   Console.   HTTP.   Auto-provision.  Security: 
   HTTP 1.1 basic/digest authentication for web setup.   MD5 for SIP authentication (RFC 2069/RFC 2617).  Interface: 
   Power input: 100-240VAC, output power: DC-12V, 1A.   WAN RJ45, 10/100M port.   LAN/PC RJ45, 10/100M port.  Certificate:
   CE.   FCC.  

VoIP SIP Gateway

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. 
 

深圳市極思維智能科技有限公司

1.WIFI + 3G/SMS touch keypad alarm system
2.Android + IOS APP application easily control, Easy to give authorization.&nbsp;
3.Free charges through WiFi network. When there is no WIFI, the host will work &nbsp;with 3G network. If there is neither WiFi nor 3G network ,then the alarm host will send alarm SMS and make alarm call. &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;4.Work with IP camera, The IP camera APP embeds in the host APP.
5.Workable with maximum 100pcs smart socket/wireless relay output to control simple home appliances.Relay outputs name and sockets name can be revised by users
6. 8 scene settings and 20 ifttt settings.&nbsp;
&nbsp; &nbsp;Advanced home automation functions for perfect smart home life.
7.Workable with RFID wireless keypad
8.20s Leaving message remotely in APP.&nbsp;
9.Information push promptly when AC power lost or recovered or detectors in low voltage
10.Self-checking windows/doors status and chime reminds if door open
11.100 wireless defense zones, Zone names can be revised by users
12. &rdquo;ding dong&rdquo; doorbell function,when someone press doorbell,the APP application will pop up push message&quot; someone is pressing your doorbell&quot; ,if user installed the YOOSEE IP camera in front of door,can open the IP camera to two way talk.
13.Can store 6 alarm call phone No.+2 alarm SMS phone No
14.Arm/alarm Delay for each defense zone. Timed arm/disarm function
15.Built-in rechargeable battery&nbsp;
16.The host support Monitoring and two-way intercom function.

 Packaging  	 standard package : 	 ①alarm host*1pcs
	②wireless motion sensor*1pcs
	③wireless door/window contact*1pcs
	④ remote controller *2pcs
	⑤ Doorbell button*1pcs
	⑥ AC power adapter*1*1pcs  

WiFi + GSM + GPRs Home Alarm System with doorbell

易數碼科技公司

 Solution :  GM8136S/GM8135S +OV9712
 Lens :  1/3&quot; Fisheye lens, f=1.44mm, F=2
 Field of vision  : Horizontal 180&deg;/Vertical 110&deg;
 Iris  : Fixed
 Sensibility  : 0.2Lux @F2(Color)/0.02Lux @F2(B/W)
 Day/Night :  Auto IR Cut filter
 Video Code  : H.264&nbsp;
 Frame Rate  : 25 fps
 Resolution  : 1280&times;720Audio : AAC/G.711，Two-way voice intercom
 WiFi standard  : IEEE802.11b/g/n
 Frequency  : 2.4GHz～2.4835 GHz
 Power port  : Micro USB
 Storage  : TF（Micro SD）up tp 128G
 Power supply  : DC5V，&lt;1A
 Maximum Power consumption  : &lt;5W
 Size :  40 x 110mm
 Weight  : 100g
 Working condition  : -10℃~+50℃@Humidity 10%~ 95%
 APP  : IPC360 &nbsp;

1.0M Pixel 180° Fish Eye IP Camera

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

This Gigabit IP phone with 6 SIP lines features an advanced design. It is equipped with the latest Broadcom's high-definition voice chip set, with a high-quality handset and a speaker. This superior IP phone is suitable for receptionists, advanced staff and department managers who have lots of calls every day.

 
  Easy operation. 
  Long lifespan. 
 

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