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Keyword "VoIP"
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Snom Snom

 Bymacht deals with Snom VoIP phones and assure good quality and best pricing.To know more details write to us.

[Related Categories: Network Communications ]
[Related Keywords: VoIP ]
  	 voip ata - 2FXS - KTA2110(2S) voip ata - 2FXS - KTA2110(2S)

Kta2110(2S) is a 2FXS voip ATA, with router /bridge function

Competitive features:
Sip protocol
For consumer or family use
2fxs ports
Ethernet ports: Two* 10/100 base-t ethernet ports,
Bridge/router functionfunctions: Out route selection(use voip or pstn),
Functions: volume adjust, auto provisioning, auto update
Sip features: Call hold/retrieve, call waiting, call transfer,3-way calling, g.711 fax, t.38 fax, mwi, do not disturb, call forwarding: Busy, no answer, unconditional
Impedance: Optional
Size:108* 72*26 (l *w *h)

 Liu Yuxiang
 Koncept Network Communication Equipment Co., Ltd.
 Room A2801~2804, Shen Fang Plaza, Ren Min Nan Rd,
 Luohu, Shenzhen, Guangdong Province,
 China
 Tel: 86-755-33387808 ext.809
 Mobile: 86-13543317817
 Fax: 86-755-33387809
 MSN: liu_yuxiang (AT) 163 (DOT) com
 Mail: lyx (AT) konceptusa (DOT) com
       lyx (AT) konceptproduct  (DOT) com

 

[Related Categories: Network Communications ]
[Related Keywords: VoIP, Ata, SIP, 2S, 2FXS, 2port ]
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USB Stick Phone for VoIP (KU2200) USB Stick Phone for VoIP (KU2200)

KU2200 is a USB memory stick with build in sound card. We put softphone such as Skype, X-lite in it so that this product can \'auto-run\' the softphone after it plugged in your PC. It do not require any installation and sound card, simply plug and dial.

[Related Categories: Network Communications ]
[Related Keywords: VoIP, USB, stick phone ]
VoIP Ata Adapter (Ipstn, 1fxs, 1LAN, 1wan) - Kta2110 VoIP Ata Adapter (Ipstn, 1fxs, 1LAN, 1wan) - Kta2110

KTA2110 is a 1FXS, 1PSTN VoIP ATA, equipped with Bridge & Router function, it could create a VPN (Virtual Private Network) connecting several computers in your family to the internet.

Additionally, it has a PSTN port which can access to PSTN line at the same time. You can shift freely between PSTN mode and VoIP one. What\'s more, when power lost, it can shift to PSTN mode automatically. Fully support Asterisk. In our promotion time, the whole sale price for 100~1000 PCS is only USD39

Competitive Features:
SIP protocol
For consumer or family use
1 FXS port, 1 PSTN (Can use for PSTN in and out to PSTN) port
Ethernet ports: TWO* 10/100 Base-T Ethernet ports,
Bridge/Router functionFunctions: Out route selection(use VoIP or PSTN),

Auto provisioning, auto new version check, auto upgrade

Functions: Caller ID transfer(VoIP and PSTN), Volume adjust

SIP features: Call hold/Retrieve, Call waiting, Call transfer, 3-way calling, G. 711 fax, MWI, Do Not Disturb, Call forwarding: Busy, No answer, Unconditional, T. 38 FAX

Impedance: Optional
Size: 108* 72*26 (L *W *H)

[Related Categories: Network Communications ]
[Related Keywords: VoIP, Ata, adapter ]
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JR-820 IP PHONE JR-820 IP PHONE
Features
Support SIP 2.0 (RFC3261) and correlative RFCs
Supprt IAX2
Codec:G.711A/u, G.7231 high/low, G.729, G.722
alternative
PoE(Power Over Ethernet),Headset
Pac king:

Inner box MEAS 310*235*75mm
G.W./unit 1.2kg
Carton MEAS 498*410*345mm
G.W. 12kg
PCS/CTN 10pcs
[Related Categories: Interphones ]
[Related Keywords: VoIP, IP, POE PHONE ]
VoIP ATA VoIP ATA
Setu ATA

Setu ATA converts VoIP networks to traditional telephony interface and vice-versa
Setu ATA has a 2 channel (2 SIP Accounts) for VoIP calls
Allows 2 Simultaneous VoIP Calls using 2 FXS Ports
It can be interfaced with PBX or GSM FCT having FXS Port
There is no need to change PBX configuration
There is no need to upgrade PBX for making IP calls
 
Setu ATA 2LL: SIP based VOIP adaptor with two FXS and One FXO lifeline ports. It has SIP based Analog Terminal Adaptor (ATA) with 2 SIP accounts, 2FXS ports, 1 FXO lifeline, 1 Wan and 1 LAN ports.
Setu ATA2S: SIP based VOIP adaptor with Two FXS ports. It is SIP based Analog Terminal Adaptor (ATA) with 2 SIP Accounts, 2 FXS ports and 1 WAN port.

Hardware Features:
Compact and Sturdy Design
LED Indications
SLIC Based Design
DSP Technology
Codec Technology
SMT Technology
Mounting Options
SLIC and SMT Design
Less Heat Generation
Dual Protection
Powder Coated Aluminum System Sub-Rack
Wall and Table Top Mounting

Software Features:
Call Forward
Call Hold
Call Waiting
Call Transfer
Intercom Call (FXS to FXS)
Call Conference
Calling Party Control (CPC)
Call Progress Tones
CLIP
Do Not Disturb (DND)
Incoming Call Routing
Phone Book
Dial Plan
SIP Accounts
DHCP Client
Echo Cancellation
FXO Life Line
NAT
PPPoE
QoS
SIP Account
STUN
Router
Voice Activity Detection (VAD)
Password Protection

Applications:
Stand-alone Application
With PBX Application
With PCO Application
Other Applications

[Related Categories: Others, Related Products ]
[Related Keywords: VoIP, Ata, Analog, Terminal, adapter, "analog terminal adapter", "VoIP PBX", PBX ]
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VoIP gateway HT-822P VoIP gateway HT-822P
2FXS and1PSTN and 1LAN
 
Specification:

1) 2 network ports(RJ45)

2) 2 protocols with SIP and H.323

3) Router funtction

4) Embedded encryption module

5) Free of charge for server's encrypt software

6) Can be customized

[Related Categories: Network Communications ]
[Related Keywords: VoIp equipment, VoIP, Gateway, 822P ]
VoIP phone EP-838 VoIP phone EP-838

1. EP-636 is a low-priced IP telephone developed by HyberTone Company himself.

2. It is compatible with H.323 V4 and SIP V2, possessing single file of English letters and single file of digital LCD displayer and achieving full dulplex hands-free call.

 

3. When it is disposed into H.323, EP-636 can cooperate with GK and GW of most H.323.

4. It also has perfect H.450 protocol to complete all telecom value-added services such as call transfer, call forward, call waiting, etc..

5. Cooperating with soft exchange system, it can totally replace traditional PBX or KEY LINE PHONE.

6. It can also be used as LAN phone or hot-line phone.

7. When it is disposed into SIP, EP-636 can cooperate with most SIP system. Besides having popular g.723 and g.729, it can use low-priced GSM voice compression algorithms.

[Related Categories: Interphones ]
[Related Keywords: Interphone, VoIP, Phone, 838 ]
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VoIP gateway HT-812P VoIP gateway HT-812P

I.HT-812P is a low-priced gateway developed by HyberTone Technology for foreign market.

 

 II. It can let users choose to use VoIP or the ordinary telephone line (PSTN) to call out. When the circuit is cut off, it will be switched to ordinary phone line automatically.

 

 III. This gateway has many advantages such as flexible configuration, smart appearance, multiple functions, and perfect tone quality. It is the first choice in converse safety, converse flexibility and market regularity.

Specification:

1) 2 network ports(RJ45)

2) 2 protocols with SIP and H.323

3) Router funtction

4) Embedded encryption module

5) Free of charge for server's encrypt software

6) Can be customized

[Related Categories: Network Communications ]
[Related Keywords: Telecommunications, Voip gateway, VoIP, Gateway, 812P ]
voip phone voip phone

Features:
1) Support SIP(RFC2543,FEC3261)
2) 64 caller ID display, 64 dialed number memories
3) Telephone director of 100 phone numbers
4) Hands-free call,automatic timing
5) Missed / dialed out / received memories, deletion and redial
6) Call decline/holding/forwarding
7) Sets phone number in short
8) Date / time display
9) PPPoE
10) Compatible with both DHCP and static IP address
11) TCP / IP (ARP / RARP, IP / ICMP, UDP / TCP / IP, RTP / RTCP)
12) TFTP and console
13) IEEE802.1P / 802.1Q/ToSu10BaseT / 100Base TX
14) DNS (Domain Name Server)
15) Three way conference
16) Call waiting / transfer
17) Half duplex hands-free call with good effect super sound effect

Network protocols:
1) SIP v1 (RFC2543), v2 (RFC3261)
2) IP / TCP / UDP / RTP / RTCP
3) IP /ICMP / ARP / RARP / SNTP
4) TFTP client / DHCP client / PPPoE client
5) NAT / DHCP server
6) Telnet / HTTP server
7) DNS client / DDNS / VLAN
8) Voice quality
9) VAD / CNG / LEC
10) Packet loss compensation
11) Adaptive jitter buffer
12) Codec
13) G.711 / G.723.1 / G.726 / G.729A / G.729B
14) DTMF function
15) In-band DTMF
16) RFC 2833 DTMF
17) SIP info
18) Phone function
19) Volume adjustment
20) Speed dial key
21) Phone book
22) Flash
23) IP assignment
24) Static IP / DHCP / PPPoE
25) NAT Traversal
26) Web browser
27) QoS
28) ToS field
29) Firmware upgrade
30) TFTP / Console / HTTP
Net Weight:1.1kg/pcs
Gross Weight: 1.2kg/box
14kg/ctn
Size:
25.5*7.6*27.5 cm(one box)
40*35.3*46 cm(one carton)
[Related Categories: Network Communications, Telecom Parts ]
[Related Keywords: voip phone, voip device, VoIP, Phone ]
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Callmaster - Call Shop Telephony Solution (Voip + Pbx ) Callmaster - Call Shop Telephony Solution (Voip + Pbx )
Specifications: Product Description: The callmaster is a hardware& software call-shop management solution that is successfully used to operate many call shops situated in countries across europe and middle east. A call-shop (also known as tele-centers, phone-shops) is a privately owned business that offers its customers the ability of performing local or international phone calls at discount/competitive rates. It can be situated in key locations in a city such as supermarkets, universities, kiosks, hotels etc Hardware and software parts of the callmaster call-shop system solution 1. Callmaster call interface device - the interface device is the link between the computer, the display units and the call carrier. It can accept and control up to 16 lines on real time and it can be set to accommodate various call service facilities(voip or from the local exchange individually or at the same time 2. Callmaster display units - assist the customers in monitoring the cost of their call/calls 3. Callmaster call shop management software - enables the call shop owner to perform all actions that are necessary to effectively operate the call-shop The main features of the callmaster call-shop system solution, among others, are the following: . Real time monitoring and control of up to 16 call booths. . Call routing through preferred service provider(normal exchange lines or voip gateway vendor - through internet). With this system a call-shop can operate either with voip or with standard analogue line from the local pstn exchange or even a simultaneous combination of the two. . Ability to efficiently control all operational and financial aspects of his business. The callmaster software provides the ability to set own call rates and tariffs as well as surcharge, provides extensive financial, statistical and billing reports/invoices thus avoiding the requirement for an external billing software or a billing process from the internet gateway vendor. . Possibility to program the system to operate with a preferred language, such as greek, english, arabic etc. . Security and back up features etc. . Very cost effective solution. The callmaster system is reliable, user friendly, extremely efficient and requires the minimum of investment thus enabling any interested party to set up, in a few hours a full working profitable call-shop business. Our product can also be complementary to the services that internet providers offer(isp's) as it can be promoted as an internet business solution to existing/new clients. This can be achieved by promoting, to interested clients, a voip callshop set up configuration thus benefiting all parties involved significantly. If you find the idea of opening a call-shop business or developing such a partnership to your interest please contact us via email for more information. A demo program can also be provided in order to appreciate how our system operates as well as its features and facilities.
[Related Categories: Others ]
[Related Keywords: Callmaster, -, Call, shop, Telephony, solution, (Voip, +, PBX, ), Callmaster, Call, shop, Telephony, solution, VoIP, PBX ]
VoIP Equipment VoIP Equipment
Specifications: - Easy to install and maintain - Extendable up to 16 ports (FXO+FXS) - 1 LAN port and 1 WAN port - Simple upgrade via TFTP - Managed via Web Browser - NAT function :Allocates up to 30 virtual IP addresses - Internet Protocol:IP,TCP,UDP,TFTP,ICMP - VoIP Protocol : H.323, H.225, H.245, RTP, RTCP,RAS - IP Mode Protocol : Static, DHCP, ADSL(PPPoE) - Voice Codec : VAD(Voice Activity Detection),CNG(Comfort Noise Generation) G.711 A-law,G.711 U-law,G.723.1,G.729A - Controled by the Web and Console - support FoIP ( Fax over Internet ProtocolCallgate Certification(s): FCC, CE
[Related Categories: Others ]
[Related Keywords: VoIP, equipment, VoIP, equipment ]
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VoIP Equipment VoIP Equipment
Specifications: - Easy to install and maintain - Extendable up to 16 ports (FXO+FXS) - 1 LAN port and 1 WAN port - Simple upgrade via TFTP - Managed via Web Browser - NAT function :Allocates up to 30 virtual IP addresses - Internet Protocol:IP,TCP,UDP,TFTP,ICMP - VoIP Protocol : H.323, H.225, H.245, RTP, RTCP,RAS - IP Mode Protocol : Static, DHCP, ADSL(PPPoE) - Voice Codec : VAD(Voice Activity Detection),CNG(Comfort Noise Generation) G.711 A-law,G.711 U-law,G.723.1,G.729A - Controled by the Web and Console - support FoIP ( Fax over Internet ProtocolCallgate Certification(s): FCC, CE
[Related Categories: Others ]
[Related Keywords: VoIP, equipment, VoIP, equipment ]
VoIP Phone VoIP Phone
Features: 1) PC to PC and PC to phone operation 2) Plug and play 3) Powered via USB port (no need for a power adaptor) 4) Ring tones and adjustable volume 5) USB 1.1 port
[Related Categories: Others ]
[Related Keywords: VoIP, Phone, VoIP, Phone ]
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VoIP Phone VoIP Phone
Features: 1) PC to PC and PC to phone operation 2) Plug and play 3) Powered via USB port (no need for a power adaptor) 4) Ring tones and adjustable volume 5) USB 1.1 port
[Related Categories: Others ]
[Related Keywords: VoIP, Phone, VoIP, Phone ]
IPCallmaster, Callshop Solution, VOIP Telephony IPCallmaster, Callshop Solution, VOIP Telephony
Specifications: The ip-CALLMASTER was developed based on the philosophy of a stand-alone callshop. This means enabling the owner to exclude unwanted web or billing hosting carriers/partners and providing him with unprecedented control and instant access to all necessary information. It efficiently manages all of the callshops activities, such as billing, tariff rating, financial and statistical reporting with reliability and security. A unique feature of the ip-CALLMASTER is that, all ip-phones can be programmed to show the call statistics through their display screen (i. E cost of current call, total cost etc). This excludes any unnecessary costs associated with call displays and simplifies the set up process. Least cost routing, multiple VOIP Providers, Real time monitoring of unlimited number of Call booths, are a few of the many other features that characterise the solution. Its simple set up, the user-friendly functionality of the software and primarily its low cost investment allows entrepreneurs to enter a lucrative business with great returns. Our solution is also suitable for existing Internet cafe owners, Internet service providers and to companies offering Carrier services/VOIP termination as it can compliment their services and assist them in expanding their profitability. Other features of the ip-CALLMASTER 1. Flexible Call routing process (Least cost, automatic or default routing) 2. The IP Phones can show on their screen all the call details to the caller during conversation 3. The system can control and monitor in real time an unlimited number of call booths 4. Ability to introduce Several VOIP Service Providers 5. Ability to set Local/National/International call rates 6. Prepaid or post-paid options as well as customer call accounts 7. Instant invoicing 8. Ability to customize the software program to a preferred language 9. Generation of various financial and statistical reports 10. Ability to set authorisation levels for each user/operator 11. Security and Back up features Please contact us at if you wish to obtain more information about ip-CALLMASTER. We welcome any questions/suggestions concerning our products and we will be happy to discuss your project, and see if we can be of any assistance.
[Related Categories: Others ]
[Related Keywords: IPCallmaster,, CallShop, Solution,, VoIP, Telephony, IPCallmaster, CallShop, solution, VoIP, Telephony ]
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IPCallmaster, Callshop Solution, VOIP Telephony IPCallmaster, Callshop Solution, VOIP Telephony
Specifications: The ip-CALLMASTER was developed based on the philosophy of a stand-alone callshop. This means enabling the owner to exclude unwanted web or billing hosting carriers/partners and providing him with unprecedented control and instant access to all necessary information. It efficiently manages all of the callshops activities, such as billing, tariff rating, financial and statistical reporting with reliability and security. A unique feature of the ip-CALLMASTER is that, all ip-phones can be programmed to show the call statistics through their display screen (i. E cost of current call, total cost etc). This excludes any unnecessary costs associated with call displays and simplifies the set up process. Least cost routing, multiple VOIP Providers, Real time monitoring of unlimited number of Call booths, are a few of the many other features that characterise the solution. Its simple set up, the user-friendly functionality of the software and primarily its low cost investment allows entrepreneurs to enter a lucrative business with great returns. Our solution is also suitable for existing Internet cafe owners, Internet service providers and to companies offering Carrier services/VOIP termination as it can compliment their services and assist them in expanding their profitability. Other features of the ip-CALLMASTER 1. Flexible Call routing process (Least cost, automatic or default routing) 2. The IP Phones can show on their screen all the call details to the caller during conversation 3. The system can control and monitor in real time an unlimited number of call booths 4. Ability to introduce Several VOIP Service Providers 5. Ability to set Local/National/International call rates 6. Prepaid or post-paid options as well as customer call accounts 7. Instant invoicing 8. Ability to customize the software program to a preferred language 9. Generation of various financial and statistical reports 10. Ability to set authorisation levels for each user/operator 11. Security and Back up features Please contact us at if you wish to obtain more information about ip-CALLMASTER. We welcome any questions/suggestions concerning our products and we will be happy to discuss your project, and see if we can be of any assistance.
[Related Categories: Others ]
[Related Keywords: IPCallmaster,, CallShop, Solution,, VoIP, Telephony, IPCallmaster, CallShop, solution, VoIP, Telephony ]
VOIP GSM Gateway VOIP GSM Gateway
MAJOR FUNCTIONS: VOIP (SIP) - GSM Conversion (SC-375) VOIP (SIP) - CDMA Conversion (SC-375C) 50 sets of LAN -> mobile Routes Setting; 50 set of Mobile -> LAN Routes Setting Voice response for setting and status (Dial in from mobile) Series connection to save bill Standard SIP Protocol (RFC2543, RFC3261) Communicates with other Gateway or PC All function can be set up via web
[Related Categories: Wireless Networking Equipment ]
[Related Keywords: VoIP, Gsm, Gateway, VoIP, Gsm, Gateway ]
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VOIP GSM Gateway VOIP GSM Gateway
MAJOR FUNCTIONS: VOIP (SIP) - GSM Conversion (SC-375) VOIP (SIP) - CDMA Conversion (SC-375C) 50 sets of LAN -> mobile Routes Setting; 50 set of Mobile -> LAN Routes Setting Voice response for setting and status (Dial in from mobile) Series connection to save bill Standard SIP Protocol (RFC2543, RFC3261) Communicates with other Gateway or PC All function can be set up via web
[Related Categories: Wireless Networking Equipment ]
[Related Keywords: VoIP, Gsm, Gateway, VoIP, Gsm, Gateway ]
VoIP Gateway (SVG6032S V1.0) VoIP Gateway (SVG6032S V1.0)
Features: 1) SIP (RFC 3261, 3262, 3264) 2) Voice Packet Encapsulation and Description Protocols RTP (RFC 1889) and SDP ] (RFC 2327) 3) DTMF in-band and out-band (RFC 2833) (SIP INFO) 4) DDNS (secure and innovative dynamic DNS service) (SIP version) 5) Subscriber/Notify method (RFC 3265) (SIP version) 6) Voice Activity Detection (VAD) and Comfort Noise Generation (CNG) 7) Silence suppression 8) Adaptive jitter buffer 9) Compensation for Loss of Packet 10) In-band or out-band DTMF relay 11) Selectable Tx/Rx gain adjustment 12) Tone generation (GR-506) and custom definable 13) Caller ID (Bell Core-GR-30, FSK, ETSI) Voice algorithms: Supports voice compression with ITU-T standard compatible FAX mode: 1) Transparent FAX mode (G3, T.30) 2) FAX auto-detection 3) FAX relay mode: ITU-T T.38 real time G3 FAX over IP protocol 4) Supports ITU-T V.21, V.27ter, V.29 and V.17 up to 14,400bps Quality of service (QoS); 1) Type-of-service (ToS) bit-tagging 2) VLAN Tagging 3) Internal queues/buffer management for voice prioritization Voice features 1) Call hold 2) Call waiting 3) Call transfer 4) 3-way call conference 5) Call forward (always, busy, no answer) 6) Speed dial 7) Polarity reversal 8) Distinctive rings 9) FAX 10) Caller ID 11) Caller ID blocking (via 3323 and 3325) 12) Voice-mail notification 13) Internal phone book and digitmap (SIP only) 14) Configuration hook-flash timing Data features: 1) Advanced routing 2) Gateway/bridge mode features 3) PPPoE features 4) Static or dynamic using DHCP client 5) Configurable DHCP free IP range for LAN interface NAT: 1) STUN (SIP) 2) Dynamic DNS (SIP) Data networking 1) MAC address (IEEE 802.3) 2) IPv4 (RFC 791) 3) ARP - Address Resolution Protocol 4) DHCP Client (RFC 2131) 5) ICMP - Internet Control Message Protocol (RFC 792) 6) TCP - Transmission Control Protocol (RFC 793) 7) UDP - User Data gram Protocol (RFC 768) 8) RTP - Real Time Protocol (RFC 1880/1890) 9) PPPoE (RFC 2516) 10) Type of Service, TOS (RFC 2030) 11) MD5 password encryption (RFC 1321) 12) Encryption of configuration file 13) Internet multicast backbone (RFC 2327) Management and provisioning: 1) Remote software upgrade and configuration 2) Upload/download via TFTP 3) Telnet/Craft CLI 4) NMS (Network Management System) based on SNMP 5) Trap 6) Username and password authentication 7) Advanced MGCP/SIP authentication 8) Debug capability star-net
[Related Categories: Wireless Networking Equipment ]
[Related Keywords: VoIP, Gateway, (SVG6032S, V1.0), VoIP, Gateway, SVG6032S ]
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