香港恒电科技有限公司

这款 PoE 交换机通过单根 Cat-5 网线，利用以太网供电 (PoE) 技术，从单一接入点同时提供电源和数据。它适用于任何 10/100/1000Mbps 链路，并提供符合行业标准的 IEEE 802.3af/at 电源。先进的自动检测算法可为 802.3af 终端设备供电。此外，该 PoE 交换机还具备自动上行链路功能，能够自动检测 PoE 需求、速度、双工模式和网线类型。操作简便，性能可靠。 PoE交换机非常适合为IP摄像头、WLAN接入点、IP电话、办公门禁系统和其他PD设备等PoE设备供电，并提供一系列高质量的产品，为不同环境下的以太网应用提供完整的解决方案。 ◆ 24个10/100Mbps PoE端口，2个10/100/1000Mbps上行链路端口 ◆ 传输距离100米，使用后可扩展至250米 ◆ 兼容 IEEE802.3AF/AT ◆ 总功率：300W ◆ 所有端口均支持 MDI/MDIX 自动翻转和自协商 ◆ 供应 24 个 10/100Mpbs 自适应高速转发数据包无丢包端口。 ◆ 每个端口最大供电功率达到 30W。 ◆ 低发热设计，安全可靠，稳定性高 ◆应用领域：无线接入点、IP摄像头、IP电话、计算机网络 

24+2 1000Mbps POE Switch 

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. 
 

Leader Team Investments Limited

 	 Smart hybrid PBX system and medium sized business Expand from 6 analogue trunks and 18 extensions, up to 288 extensions with expansion unit Cost effective installation and operation Built in applications include call centre solution, mobile solution and voicemail Easy evolution from analogue to IP communications  

PABX System

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

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

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

   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.

   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 

   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

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)

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

佛山市容讯科技有限公司

Model:
   CDX-TS208.   CDX-TS308.   CDX-TS416.  Main Features:
   Metal cover.   Switch power (optional).   Backup battery interface (optional).   Transfer caller ID (optional).   Intercom caller ID (optional).   Auto-attendant (DISA:10s one paragraphs).   Auto-attendant (DISA:20s one paragraphs) (optional).   Auto/Operator mode.   Auto IP dialing (optional).   External music input on holding (optional).   Doorbell &amp; door open function (optional).   11 chord melody music selectable, auto select daily ((optional).   CO line transfer to CO line (optional).   Wireless terminal board (optional).  Function:
   Multiple communication line.   Normal transform power.   Power failure transfer.   External line caller ID.   Internal music on holding.   Remote programming.   Memory protect while power off.   System password.   Multiple dial-out mode.   External line group.   Call duration control.   Call pickup.   Outgoing call transfer.   Call transfer.   Call forwarding.   Flexible coding (change ext.No.).   Do not disturb.   Automatic Ext. reporting.   Flash time choice.   Ringing assignment.   Call restriction.   Three way conference.   Call all extensions.   Specific CO line dialing.   Ring identify.   CO line booking.   CO line reservation.   Operator setting.   CO line monitor.  Packing:
   Weight: 1.7~2.0 KGS/PC.   Quantity: 5pcs in one carton.   Giftbox size: 40.7x28.8x8.3cm.   Carton size: 43.5x42x31.5cm.  

Metal PABX Telephone Switch System

Function:   Auto-Attendant(DISA: 20s three paragraphs) (CS+ series).   Auto-Attendant(DISA-12s one paragraph) (CS+ series).   Operator setting.   External line caller ID.   Transfer Caller ID (CS+ series).   Intercom Caller ID (CS+ series).   Call transfer.   Outgoing call transfer.   C. O. Line Transfer to C. O. Line.   Call pickup.   Call forwarding.   Call all extensions.   Three way conference.   Auto IP Dialing (CS+ series).   Specific CO line dialing.   CO line booking.   CO line reservation.   Ringing assignment.   Do not disturb.   Multiple dial-out modes.   Capacity: 4 Co lines, 24/32 extensionsperator /Auto-attendant.  Feature:
   Exts. Groups.   External Line Group.   Door Phone Interface (optional for CS+ series).   Door opener (optional for CS+ series).   Internal music on holding.   External Music Input on Holding.   11 Chord Melody Music Selectable (CS+ series).   Call duration control.   Call restriction.   Ring identify.   Outgoing Message Recording.   C. O. Line &amp; Exts Monitor.   Flexible coding (change ext. No. ).   Automatic Ext. Reporting.   System CRBT.   Day/Night Mode (Auto/Manual).   Multiple communication lines.   Flash time choice.   Remote programming.   System Password.   Power failure transfer.   Memory protect while power off.   Lightning Protection.   Backup Battery Interface (optional).   Normal transform power.   Switch Power (optional).  Packing:
   Weight: 2.3~4.2KGS/PC.   Quantity: 5PCS in one carton.   Gift box size: 45&times; 29&times; 11cm.   Carton size: 56&times; 47&times; 31.5cm.  

Central Telephony System

 Features:    Plastic cover.   Switch power (optional).   Backup battery interface (optional).   Transfer caller ID (optional).   Intercom caller ID (optional).   Auto-attendant (DISA: 10s one paragraphs).   Auto-attendant (DISA: 20s one paragraphs) (optional).   Auto-attendant (DISA: 60s or 90s four paragraphs) (optional).   Auto/Operator mode.   Auto IP dialing (optional).   External music input on holding.   Doorbell &amp; door open function (optional).   11 chord melody music selectable, auto select daily ((optional).   CO line transfer to CO line (optional).  Function:
   Multiple communication line.   Normal transform power.   Power failure transfer.   External line caller ID.   Internal music on holding.   Remote programming.   Memory protect while power off.   System password.   Multiple dial-out mode.   External line group.   Call duration control.   Call pickup.   Outgoing call transfer.   Call transfer.   Call forwarding.   Flexible coding (change ext. No. ).   Do not disturb.   Automatic Ext. Reporting.   Flash time choice.   Ringing assignment.   Call restriction.   Three way conference.   Call all extensions.   Specific CO line dialing.   Ring identify.   CO line booking.   CO line reservation.   Operator setting.   CO line monitor.  Packing Details:
   Weight: 1.7~2.0 KGS/PC.   Quantity: 5PCS in one carton.   Giftbox size: 40.7x28.8x8.3cm.   Carton size: 43.5x42x31.5cm.  

Analog Telephone Switch PABX System

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.  

VoIP Phone

YEASTAR TECHNOLOGY CO., LIMITED

NeoGate TG400 is a compact 4 channels VoIP GSM gateway that connects GSM network with VoIP. It is the ideal product for small and medium sized companies with heavy demands of calls to mobile networks. The cost-saving solution makes connection cheaper and efficient

Benefits
1) Cost savings - save phone bills tremendously with mobile-to-mobile calls and LCR
2) Back up - work as a cost-effective backup when the landline network goes down
3) Easy to install - everything can be easily set up in the Web based management interface
4) Easy to integrate - high compatibility with major IP PBX and soft switch brands

Specification   Number of GSM channels (max): 4   Network type: 850/900/1800/1900MHz   Protocol: SIP (RFC3261), IAX2   Transport protocol: UDP, TCP, TLS, SRTP   DTMF: RFC2833, SIP INFO, In-band   Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.   Echo cancellation: ITU-T G.168 LEC   LAN: 1 (10/100Mbps)   Network: static IP, DHCP client, firewall, VLAN, DDNS   QoS, open VPN   NAT traversal: static NAT, STUN   Size: 213x160x44mm   Power supply: AC 100-240V 50/60Hz 0.5A MAX   Operation range: 0&deg;to 40&deg;C, 32&deg; to 104&deg; F   Storage range: -20&deg; to 65&deg;C, -4&deg; to 149&deg; F   Humidity: 10-90% non-condensing  
Features   SIP server and SIP trunk supported   SIP peer mode supported   Calling type: VoIP to GSM, GSM to VoIP   GSM ports group manage, VoIP trunk group   Incoming /outgoing routing rules   SMS sending and receiving   Send bulk SMS   Gain adjustment   USSD   PIN modify   Carrier selection: auto/manual   Balance alarm   Caller ID/CLIR   Black list   Hotline   Call duration limitation   Call transfer   Call back   Call status display   Call detail record (CDR)   Call progress tone generation   Call duration limitation for SIM card/single call   LCR (least cost routing)   Top voice quality (EFR super sound)   SIP response code switch   Open API for SMS and USSD   Real open api protocol (based on asterisk)   IP blacklist   Network attack alert   System logs   Web based configuration  

VoIP-GSM Gateway

深圳市源创通信技术有限公司

The SinoV-S500 is an analog GSM cellular VoIP gateway that helps directly interconnect a SIP-supporting VoIP network with GSM networks. It features a strong housing to offer reliable protection to the internal components. Well engineered, this device ensures excellent performance.
 
  4/8 channels: 850/900/1,800/1,900MHz quad-band GSM cellular network. 
  SMS receiving/batch sending. 
  AT command compatible. 
  Support SIP registration and peer-to-peer. 
  Encryption: RC4, VOS, SinoV private, etc. 
  Hotline, phone book, speed dial, and dial plan. 
  Update firmware by local or Internet. 
  Field proven compatible with asterisk-based soft switches. 
 

GSM VoIP Gateway

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

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. 
 

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