进达电子制品有限公司

 	 Maximum of 6 VoIP accounts - IPv4 - WAN port and LAN port - POE - Wideband G.722 Codec 	 Server Compliance Asterisk / Broadsoft / Alcatel / NEC / Avaya / Aastra&nbsp; 	 Dual power supplied Support 1.8GHz (EU) and DECT 6.0 (US) frequency band  

VoIP Dect Phone

&bull; 3&quot; monochrome LCD
&bull; Multi-sip account
&bull; PoE 802.11af
&bull; VLAN, SRTP, HTTPS
&bull; 8 one-touch speed-dial keys
&nbsp;

VoIP Phone

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

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

Mercury Corporation

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

LTE CPE / LTE to WiFi 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

深圳市沃宇科技有限公司

   VG1X-8 gateway is a perfect product for the telecom carriers to provide the residential telephony and fax services through the broadband networks such as FTTB, HFC, and ADSL. VG series gateway can be connected to the traditional PBXs to provide the voice solutions for enterprises, government, and schools. VG Series Gateway can also serve as a remote SIP terminal for IP-PBX solutions. VG Series Gateway is powerful hardware platform provides sufficient power for IP packetization, voice compression and echo cancellation even when the voice traffic is at the peak.   VG series SIP gateway is based on the embedded Linux operating system. With its powerful hardware platform and flexible software design VG Series Gateway can meet various voice application requirements.  Key feature:
   Line configuration 4 or 8 FXS analog ports; 4 FXO analog ports, mixed 4FXS/FXO analog port.   Voice processing voice codec: G.711, G.729A, G.723.1, GSM, iLBC.   End user features caller ID, call forwarding, call waiting, hotline, prefix dialing, DND.   Fax processing T.30, T.38 fax relay.   Billing radius interface; polarity reverse.   Configuration methods web based user interface; text based configuration.   Remote provisioning HTTP/web, auto provision, remote configuration, software upgrade, collects log files.  Signaling:
   Off hook detectio FXS loop start, DDI (Hong Kong).   VoIP protocol SIP (RFC3261), MGCP (RFC2705) 3GPP TS 24.228, TS 24.229.   DTMF in-band: RFC2833, INFO (SIP).  Other special feature:
   Auto dial dtMF mode.   Dual telephone number FXS port can have two number for incoming calls (PSTN and extension).   Support primary and backup the gateway can be configured and controlled by multiple soft switches.   Fashion ringing tones Can download and store 20 music files.  Hardware:
   Maximum user line length 3000 meters.   Ringer equivalence number 5 for short loop (300 meters), 3 for medium loop (1500 meters).   Ringing voltage 60V RMS.   Line interface RJ11.   Ethernet interface connector RJ-45 10/100 Base-T.   Internal memory 32MB.   Flash memory 8MB.   Input voltage 110～240V AC, 50～60Hz, input current 1A max.   CPU Motorola Power Quicc 852.   DSP TI 5509A.   Power consumption: 5V/3A DC (A AC/DC convector is provided).   Dimension (H x W x D): 195 x 118 x 24 millimeters.   Weight: 0.65kg.  

8 FXS/FXO Port VoIP Gateway

HUNG TAT INT'L (HK) LTD

This digital telephone answering machine features a metal keypad with a laser-etched backlight option.  The&nbsp;large-scale phone book stores up to 200 contact&nbsp;entries.  This telephone answering machine is a&nbsp;modern office assistant in&nbsp;busy companies.
 
  Handsfree speaker. 
  CID log:&nbsp;30 records. 
  Melody:&nbsp;4 monotones, 5 polytones, 9 optional polyphonic. 
  Intelligent low radiation. 
 

Digital Telephone Answering Machine

Shenzhen Cathay InfoTech Co., Ltd.

This phone is designed with a large monitor to provide touch-screen functions. The touch-screen design provides one-touch communication for patients and aged persons in hospitals or nursing homes. A video camera is set on top of the screen.
 
  Easy operation. 
  Available in various designs. 
 

TV 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

   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

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

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


 
 


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

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

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

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

Vogtec (H.K.) Co., Limited

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

VoIP Desktop

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

VoIP Over LTE and WiFi 2.4 / 5G Desktop Phone 

   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 

 	 NeoGate TE100 is a single port VoIP E1/T1/J1 gateway (VoIP to E1/T1/J1, and E1/T1/J1 to VoIP) that supports up to 30 concurrent calls. It&rsquo;s designed to bridge the gap between E1/T1/J1 and VoIP networks. TE100 offers SMBs cost effective additions to an legacy telephone system to bring the true benefits of VoIP. Integrating TE100 into an existing network will allow for inexpensive communication via SIP trunking. Also, it could connect VoIP systems with E1/T1/J1 service from legacy carriers  Specification: 	 E1/T1/J1 Port: 1 (Support PRI, MFC R2, SS7) 	 Protocol: SIP (RFC3261) 	 Transport Protocol: UDP, TCP, TLS, SRTP 	 DTMF: RFC2833, SIP INFO, In-band 	 Codec: G.711 a/u-law, G.722, G.726, G.729 A, GSM 	 LAN: 10/100 Mbps 	 WAN: 10/100 Mbps 	 Size: 213 x 160 x 44mm 	 Power Supply: AC 100~240V, 50~60Hz (DC 12V, 1A)  Features: 	 Trunk Support 	 Call Routing Rules 	 Automatic appending and stripping of digits to dialed numbers 	 Caller ID name and number support 	 FAX support 	 Blacklist 	 Firewall 	 DDNS Support 	 Backup And Restore 	 Easy to install  Benefits: 	 Cost-effective Call Routing 	 Easy-to-navigate GUI 	 Simple Installation 	 Easy to integrate 	 Low power consumption for your green office  Applications: 	 Connect legacy PBX systems to VoIP services 	 Connect legacy PBX systems to remote sites over private VoIP links 	 Connect IP PBX systems to legacy TDM services 	 Phased transition from Legacy PBX to IP PBX  

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

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. 
 

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