Zhuhai Hiwink Technology Co., LTD.

IP65 Waterproof: Ensures durability against rain and dust, ideal for outdoor use.Night Vision: Provides clear visibility even in low light conditions.Full Duplex Audio: Offers clear two-way communication for smooth interactions.Multi-language Support: Accommodates various user languages for global usability.Seamless 4-in-1 Security: Integrates multiple security features for comprehensive protection.apartment door intercom outdoor intercom ip door phone internet intercom system intercom phone with door release video doorbell with fingerprint unlock ABS material door intercom ROHS certified doorbell ONVIF network intercom FCC certified doorphone video doorphone system Tuya application intercom WiFi network doorbell IP network intercom

IP Video Intercom

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

风青实业股份有限公司

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

SIP Phone

Auraland Electronics Co Ltd

Feature is a Internet Telephone.   3 VolP accounts, hotline   Call waiting, call transfer, call forward, Hold, mute, flash, auto-answer, redial, 3-way conference, DND, speed dial   Phonebook with 140 groups memory   Volumn adjustment, ring tone selection, tone scheme, System log, DDNS function  

SIP/VoIP Internet Telephone

Fanvil Technology Co., Ltd.

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. 
 

IP Phone

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

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

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 

Eware Networks Limited

 	 Features Yealink HD Voice 	 Wall mountable  

Yealink SIP-T21P E2 Entry-Level IP Phone with 2 Lines & HD Voice

Nuzon Tech. Inc.

This two-way communication auto-dialer comes with a built-in main console. The LED display shows 2 lines clearly. Maximum 63 sensors can be supported by this digital dialer.
 
  Frequency: 303MHz, 433MHz. 
  With an AC adapter. 
 

Communication Auto-Dialer

   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

Mercury Corporation

It&#39;s QoS Function for Telecom Phone VoIP Service

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

 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 

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

LTE CPE / LTE to WiFi Gateway

    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  

VoIP-PRI Gateway (VoIP-E1/T1/J1)

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

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

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

   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

   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

   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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