Shanghai Qingbao  Robot Co., Ltd.

牙牙精灵 IP 机器人，作为专为儿童医疗场景设计的 IP 形象机器人，牙牙精灵以可爱造型与智能交互为核心，有效降低儿童就医恐惧、提升诊疗体验。它兼具三大核心价值：以拟人化形象强化医院温馨感，建立儿童信任；通过自动化导览与问答减轻医护负担；针对儿童牙科、儿科定制功能，破解就医流程痛点。在应用场景上，牙牙精灵适配多元空间：医院入口处化身 “智能迎宾”，以魔法棒灯光闪烁、“欢迎来到健康小屋” 语音互动吸引儿童注意力；人流量大的过道里，它能按预设路线导航至诊室，同步讲解护齿知识，还会做眨眼、歪头等动作引导儿童跟随；户外如地铁站出口、诊所路口，它循环播放 “爱护牙齿，找牙牙精灵哦～”，触摸脸颊显红晕，助力品牌传播与引流。功能上，牙牙精灵亮点十足：支持自定义儿童口语化语音库，500 万像素摄像头可识别儿童表情并触发安抚语音；拥有 10 + 种可爱肢体动作，搭配 27 寸大屏播放动画，还能通过 “刷牙小测试” 等游戏化问答科普护齿知识。依托智能移动底盘，它续航超 8 小时，支持激光 SLAM 导航，可精准指引挂号窗口、诊室位置，目前已在三甲医院、数字化口腔医学中心落地，成为备受小朋友喜爱的 “医疗小帮手”。 

QINGBAO 小精灵机器人

深圳市高斯贝尔家居智能电子有限公司

2560*1440 live video Dual-band wireless transmission RJ45 connection provides greater stability Remote view with iPhone/Android device Pan: 355°Tilt: 60° Two way audio Support photo taking and video recording 8meters night vision range for night surveillance Support 256GB SD card/Cloud recording optional 

T5846JEB 4.0MP Smart Pan/Tilt IP Camera

2560*1440 live video Dual-band wireless transmission Remote view with iPhone/Android device Pan: 352 ° Tilt: 55 ° Two way audio Support photo taking and video recording 10meters night vision range for night surveillance Human detection Motion detection Support 256GB SD card/Cloud recording optional 

T5842JDB Dual-band Pan/Tilt IP Camera

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

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. 
 

深圳市沃宇科技有限公司

   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

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

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

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

 Specification 
 Number of Telephony Ports: &nbsp;24
 Phone Interface:&nbsp; 24 x RJ11 FXS ports plus 1 x 50 pin
Telco connector
 Protocol: &nbsp;SIP (RFC3261), IAX2
 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.729a, GSM,
ADPCM, Speex
 Echo Cancellation:&nbsp; ITU-T G.168 LEC
 FAX over IP:&nbsp; T.38 and pass-through

 Signaling : Kewl Start, Loop Start
 Caller ID : BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
 Disconnect Method : Busy Tone, Polarity Reversal

 LAN: &nbsp;1 (10/100Mbps)
 Network: &nbsp;Static IP, DHCP Client, Firewall, VLAN, DDNS,&nbsp;
QoS, OpenVPN
 NAT Traversal: &nbsp;Static NAT, STUN

 Size: &nbsp;440 (L) mm x 250 (W) mm x 24 (H) mm
 Power Supply: &nbsp;AC 100~240V/50~60Hz&nbsp;

 Operation Range: &nbsp;0&deg; to 50&deg;C, 32&deg; to 104&deg; F
 Storage Range: &nbsp;-20&deg; to 65&deg;C, -4&deg; to 149&deg; F
 Humidity: &nbsp;10-90% non-condensing

 Features
3-way Conference 
 Attended Transfer 
 Black List 
 Blind Transfer 
 Call Detail Record (CDR) 
 Caller ID Display or Block 
 Call Forward: No Answer, Busy, All 
 Call Waiting 
 Call Status Display 
 Do Not Disturb 
 IP Blacklist 
 Network Attack Alert 
 Paging 
 RADIUS Accounting &amp; Login 
 Real Open API Protocol (Based on Asterisk) 
 Voice Prompt 
 SIP Server and SIP Trunk supported 
 SIP Peer Mode supported 
 SNMP* 
 TR-069* 
 System Logs 
 Web based configuration 
&nbsp;

Analog VoIP Gateway

 Specification 
 Number of Telephony Ports: &nbsp;8
 Phone Interface:&nbsp; 8 x RJ11 FXS ports
 Protocol: &nbsp;SIP (RFC3261), IAX2
 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.729a, GSM,
ADPCM, Speex
 Echo Cancellation:&nbsp; ITU-T G.168 LEC
 FAX over IP:&nbsp; T.38 and pass-through
 Signaling : Kewl Start, Loop Start
 Caller ID : BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
 Disconnect Method : Busy Tone, Polarity Reversal

 LAN: &nbsp;1 (10/100Mbps)
 Network: &nbsp;Static IP, DHCP Client, Firewall, VLAN, DDNS,&nbsp;
QoS, OpenVPN
 NAT Traversal: &nbsp;Static NAT, STUN

 Size: &nbsp;200 (L) mm &times; 137 (W) mm &times; 25.6 (H) mm
 Power Supply: 12V 1.5A

 Operation Range: &nbsp;0&deg; to 50&deg;C, 32&deg; to 104&deg; F
 Storage Range: &nbsp;-20&deg; to 65&deg;C, -4&deg; to 149&deg; F
 Humidity: &nbsp;10-90% non-condensing

 Features
3-way Conference 
 Attended Transfer 
 Black List 
 Blind Transfer 
 Call Detail Record (CDR) 
 Caller ID Display or Block 
 Call Forward: No Answer, Busy, All 
 Call Waiting 
 Call Status Display 
 Do Not Disturb 
 IP Blacklist 
 Network Attack Alert 
 Paging 
 RADIUS Accounting &amp; Login 
 Real Open API Protocol (Based on Asterisk) 
 Voice Prompt 
 SIP Server and SIP Trunk supported 
 SIP Peer Mode supported 
 SNMP* 
 TR-069* 
 System Logs 
 Web based configuration 

 Specification 
 Number of Telephony Ports: &nbsp;4
 Phone Interface:&nbsp; 4 x RJ11 FXS ports
 Protocol: &nbsp;SIP (RFC3261), IAX2
 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.729a, GSM,
ADPCM, Speex
 Echo Cancellation:&nbsp; ITU-T G.168 LEC
 FAX over IP:&nbsp; T.38 and pass-through

 Signaling : Kewl Start, Loop Start
 Caller ID : BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
 Disconnect Method : Busy Tone, Polarity Reversal

 LAN: &nbsp;1 (10/100Mbps)
 Network: &nbsp;Static IP, DHCP Client, Firewall, VLAN, DDNS,&nbsp;
QoS, OpenVPN
 NAT Traversal: &nbsp;Static NAT, STUN

 Size: &nbsp;200 (L) mm &times; 137 (W) mm &times; 25.6 (H) mm
 Power Supply: 12V 1.5A

 Operation Range: &nbsp;0&deg; to 40&deg;C, 32&deg; to 104&deg; F
 Storage Range: &nbsp;-20&deg; to 65&deg;C, -4&deg; to 149&deg; F
 Humidity: &nbsp;10-90% non-condensing

 Features
-way Conference 
 Attended Transfer 
 Black List 
 Blind Transfer 
 Call Detail Record (CDR) 
 Caller ID Display or Block 
 Call Forward: No Answer, Busy, All 
 Call Waiting 
 Call Status Display 
 Do Not Disturb 
 IP Blacklist 
 Network Attack Alert 
 Paging 
 RADIUS Accounting &amp; Login 
 Real Open API Protocol (Based on Asterisk) 
 Voice Prompt 
 SIP Server and SIP Trunk supported 
 SIP Peer Mode supported 
 SNMP* 
 TR-069* 
 System Logs 
 Web based configuration 

 Specification 
 Number of Telephony Ports: &nbsp;16
 Phone Interface:&nbsp; 16 x RJ11 FXS ports plus 1 x 50 pin
Telco connector
 Protocol: &nbsp;SIP (RFC3261), IAX2
 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.729a, GSM,
ADPCM, Speex
 Echo Cancellation:&nbsp; ITU-T G.168 LEC
 FAX over IP:&nbsp; T.38 and pass-through

 Signaling : Kewl Start, Loop Start
 Caller ID : BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
 Disconnect Method : Busy Tone, Polarity Reversal

 LAN: &nbsp;1 (10/100Mbps)
 Network: &nbsp;Static IP, DHCP Client, Firewall, VLAN, DDNS,&nbsp;
QoS, OpenVPN
 NAT Traversal: &nbsp;Static NAT, STUN

 Size: &nbsp;440 (L) mm x 250 (W) mm x 24 (H) mm
 Power Supply: &nbsp;AC 100~240V/50~60Hz&nbsp;

 Operation Range: &nbsp;0&deg; to 50&deg;C, 32&deg; to 104&deg; F
 Storage Range: &nbsp;-20&deg; to 65&deg;C, -4&deg; to 149&deg; F
 Humidity: &nbsp;10-90% non-condensing

 Features
3-way Conference 
 Attended Transfer 
 Black List 
 Blind Transfer 
 Call Detail Record (CDR) 
 Caller ID Display or Block 
 Call Forward: No Answer, Busy, All 
 Call Waiting 
 Call Status Display 
 Do Not Disturb 
 IP Blacklist 
 Network Attack Alert 
 Paging 
 RADIUS Accounting &amp; Login 
 Real Open API Protocol (Based on Asterisk) 
 Voice Prompt 
 SIP Server and SIP Trunk supported 
 SIP Peer Mode supported 
 SNMP* 
 TR-069* 
 System Logs 
 Web based configuration 

   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.  

VoIP Gateway

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

&nbsp;Specification
Number of GSM Channels (Max):&nbsp;16
Antenna Splitter:&nbsp;4 in 1
Network Type:&nbsp;850/900/1800/1900MHz&nbsp;
Protocol:&nbsp;SIP (RFC3261), IAX2
Transport Protocol:&nbsp;UDP, TCP, TLS, SRTP
DTMF:&nbsp;RFC2833, SIP INFO, In-band
Code:&nbsp;G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation:&nbsp;ITU-T G.168 LEC

LAN:&nbsp;1 (10/100Mbps)
Network:&nbsp;Static IP, DHCP Client, Firewall, VLAN, DDNS,&nbsp;
QoS, OpenVPN
NAT Traversal:&nbsp;Static NAT, STUN

Size:&nbsp;440x280x44 mm
Power Supply:&nbsp;AC 100-240V 50/60Hz

Operation Range:&nbsp;0&deg;C to 40&deg;C, 32&deg;F to 104&deg; F
Storage Range:&nbsp;-20&deg;C to 65&deg;C, -4&deg;F to 149&deg; F
Humidity:&nbsp;10-90% non-condensing

VoIP-GSM Gateway

   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

   Use the ALLO GSM card to bridge calls from any Mobile GSM Network to VOIP in order to reduce mobile call cost or create new connections. Our cards are used by any size company who have employees on the road or want to create extensions in remote locations. Your PBX will become mobile and extensions roaming   ALLO drivers are available for Linux 2.4X and 2.6X Kernels for easy installation with any Open Source projects such as Asterisk, Elastix, Trixbox and FreeSwitch. ALLO GSM cards have FRONT PLATE SIM LOAD technology which allows the user to insert/remove the SIM cards without opening the computer chassis  Key Features
   Echo Cancellation for Digital audio quality   User can modify IMEI and PIN   1, 2 or 4 GSM modules   Quad-Band GSM 850/900/1800/1900 MHz   Send/Receive SMS in text mode, AT command ready   Compatible with Asterisk/FreeSwitch/Elastix/TrixBox based applications   Front Plate SIM Load Technology (switch cards in a few seconds)   GSM modules are AT &amp; T certificated   USSD, CLIP, CLIR supported   DTMF detection ready   Up to 4 Simultaneous GSM Calls   Support external antenna   Uses GSM channel driver   Bi-directional calls (Inbound and Outbound calls)   Fast and easy installation   Available in both PCI and PCIe   GPRS data connections   4 LEDs (network state indicators)   Suitable for 3.3 volts and 5.0 volts 32 bit PCI slots   Operation temperature: 0&deg;C to 50&deg;C   Storage temperature: -40&deg;C to 125&deg;C   Includes a 2 year warranty against hardware defects   30 day no hassle money back guarantee   FREE Email support included   Phone Support up to 45 minutes by appointment  

GSM Card for Asterisk PCI and PCIe

   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 

联合网方案有限公司

  MultiTech Hybrid VoIP Gateways (8 Ports).  Support DID/DDI, FXS, FXO, E&amp;M, T1 and E1 interfaces.  Programmable DID/FXS/FXO/E&amp;M Port.  

MultiVOIP(R)

   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.  

SIP Phone

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


 
 


   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

深圳市宝安高科电子有限公司

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.  

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