Proolin voip iad / pl-311 voice gateway is an internet based one port voice gateway. Pl-311 adapts multi voice control protocols and voice compression codec to directly convert analog voice into ip packet for internet transport Pl-311 one port voice gateway supports sip protocols, offers two 10/100mbps ethernet interface , one rj11 telephone interface and one fxo port. Therefore, the original analog telephone can dial ip phone , it is compatible with various ippbx and voip voice gateway to provide broadband ip voice service
Procuts name: voip gateway Product model: pl-404 1. Support sip protocal, ieee 802.3 10 base t 2. Support g.711a/u, g.723.1, g.729 /a/b/ab and gsm610 codec 3. Support dhcp, pppoe dns telnet; http, tftp, ftp upgrade 4. Sopport tcp/ip, rtp rtcp, vad/cng 5. Support bridge and router model 6. 1 wan, 1 lan; 4 fxs port, More...
Voip gateway 1, Not need to register SIP PROXY, supports Point-to-Point communication (call the other partyí»s IP address directly); 2, Not need to connect any network cable, supports mutual communication between two FXS ports; 3, Three SIP PROXY and accounts registered at the same time for each FXS port (so total is 6 user names), one SIP PROXY offline, other one can replace it to work timely; 4, Built-in Router; 5. With reverse-polarity signal, billing function supported; 6. Support Voice-announcing IP address and operating by hand or by web page; 7, Support SIP 2.0 (RFC3261), TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, ARP/RARP, DNS, DHCP (both client and server), NTP, PPPoE, STUN, TFTP; 8, Powerful digital signal processing (DSP) to ensure superb audio quality; 9, Advanced adaptive jitter control and packet loss concealment technology; 10, Support various vocoders including G.711 (a-law and u-law), G.723.1 (5.3K/6.3K), G.726, G.729A/B and iLBC; 11, Support Caller ID/Name display or block, Hold, Call Waiting, Flash, Call Transfer (blind/consultative ); 12, Call forward, in-band DTMF, Dial Plans; 13, Support call waiting caller ID; 14, Support fax pass through and T.38 (pending); 15, Support silence suppression, VAD (Voice Activity Detection), CNG (Comfort Nois Generation); 16, Line echo cancellation (G.168), and AGC (Automatic Gain Control); 17, Support standard encryption and autherntication (DIGEST using Md5 andMD5-sess); 18, Support for Layer 2 (802.1q VLAN, 82.1P) and Layer 3 Qos (Tos, Diffserv, MPLS); 19, Support automated NAT traversal without manual manipulaton of firewall/NAT; 20, Support device configuration via built-in IVR, web browser or downloading encrypted configuration file through TFTP or HTTP; 21, Support firmware upgrade via TFTP or HTTP; 22, Ultra compact (wallet size) and lightweight design, great companion for travelers; 23, Desk/Wall mounts. Interface: WAN: 1 unit, connect to XDSL/Cable Modem or Network Switch or Router LAN: 1 unit, connect to PC or other network equipment FXS: 2 units, connect to Normal phone, DECT/Cordless phone, Electrograph, Switch etc Power: 1unit, connect to Power adapter: 12V DC---1A Products
The S400/S200 VoIP Gateway is fully H.323/SIP standard compliant residential gateway that provides a total solution for integrating voice-data network and PSTN. By simple installation, this revolutionary compact voice over IP (VoIP) gateway could be configured as a 2/4 FXS/FXO VoIP Gateway which provides voice connectivity over the IP network and to the Public Switched Telephone Network (PSTN).This Soundwin Gateway is equipped with a four port Ethernet switch and built-in NAT router function that provides Internet access using only one IP address. Besides, it provides high voice quality and optimized packet voice streaming over managed and public (Internet) IP networks. 300x165x30 1500g
Voip gateway Product characteristics: 1. Standard sip 2. 0 terminal product, powerful applicability, and best for global popularizing; support sip 2. 0, tcp / udp / ip, rtp / rtcp, icmp, arp / rarp, ntp, pppoe, tftp, http, etc; support g. 711 (alaw and u-law) , g. 723. 1 (5. 3k / 6. 3k) , g. 723 (40k / 32k / 24k / 16k) , g. 728, g. 729a / b, ilbc; 2. The powerful dsp digital audio processor to keep a wonderful audio-quality; inside dns, router, nat, dmz, the gateway and dhcp (the client end and the server end) ; jitter control adn packet loss concealment technology; 3. Support incoming call on show, restricting, and holding, disconnection, call transfer, call divert, dialing project, etc. Support conference call; 4. Supports vad, cng, agc, the echo eliminates (g. 168) and the static sound suppression; 5. Support layer 2 (802. 1q vlan, 802. 1p) and layer 3 (qos, , tos diffserv, mpls) 6. Powerful long-distance tftp or http manages; to dispose through the browser or the voice hint; 7. Nat automatic penetration; 8. Support passing through and t. 38 fax; 9. Good for the environment, according to the international electromagnetism radicalizations standard; 10. Select high quality material, original electrocircuit design, and no electric current noise, work steadily; 11. Porcelain and legerity design, let you feel the full-bodied appetency of the product besides expediency schlepping. Ata (sip2. 0) yes Router / bridge / dhcp yes Ethernet port 1wan / 1lan (rj45) Fxs 1fxs Fxo pstn pass-through Exit yes Weight 0. 38kg Active status temperature 0~40 Power source adapter ac in: 100v-240v dc out: + 9v / 600ma Use power:
Voip gateway Ieee 802.3 10 base tsupport dhcp, pppoe dns telnet Support g.711a/ug.723.1, g.729 /a/b/ab and gsm610 voice codec Sip, h.323 v4 protocalsip rfc3261support http, https, ftp upgrade Tcp/ip: internet protocalsupport rtp rtcp protocal Support vad/cngone wan, one lan, support router Call forward (always, busy, no answer), call waiting and per call-waiting blocking Caller number/name display caller id /block outgoing caller-id/reject anonymous call A voice activity detection (vad) facsimile protocol(option): t.30/t.38 Voip speed dial, voip digital mapper area ring tone supportdnd(do not disturb) mwi G configurable/adaptive jitter buffer sizesupport vlan protocalsupport web, ivr to configur
VoIP gateway SIP supported FXS/FXO Gateway
The ip pbx-01\02\04\08 is a complete asterisk appliance with one xo or fs module .Tt is an embedded open source linux system with built-in sip/iax2 proxy server and nat funtiongs. Tt provides a solid, uniform platform for traditional pstn communications as well voip communications. Ip pbx-01\02\04\08 provides a cost-saving solution on their telecommunication date needs .With ip pbx-01, company with brach offices in different countries can be easily combined together to work like a virtual single office through internet. 1.Open soure asterisk 2.Uclinux operation system 3.Configurable ivr meue 4.Over 50-100 avilable sip/iax2 extensions for smb use 5.20 concurrent calls 7.Call forward , call waiting , call trandfer (blind transfer/attendr transfer), call pickup/ call parking , call queues, ring group call detail record call routing 8.Conference room 9.Password protect for conference room 10.Follow me 11.Music on hold 13.Sip trunk, iax2 trunk , pstn analog trunk 14.Configuer via web interface 15.Code:g.711u/a, g.729, gsm, speek, g.726 16.Full ssh access Box
The ht-322 is designed as a compact, high performance, and low cost fxo gateway. The fxo detection is optimized to avoid the hold up of the pstn line when the other party is disconnected. This has been one of the key issues in the design of fxo gateway. The incoming pstn caller id is also transmitted to the voip user for more user friendly operation. The ht-322 is a full featured fxo gateway and is designed for easy installation and configuration. It is an ideal solution for voip to pstn termination in both sme and soho environment.
Voip Gateway IAD series to meet the NGN requirement of Telecom carrier, Internet Service Provider, SMB/SME users and home users.It is able to provide safe and high-efficiency voice services, with all kinds of protocols and functions.
The ht-912 is designed as a compact, high performance, and low cost voip analog terminal adapter (fxs gateway). It comes with one fxs port to interface with a traditional analog phone set or a pbx trunk line for voip communications. It bundles with lots of features to meet the demand in various network environments. It is an ideal low cost voip solution for travelers and soho users.
The ht-812p is designed as a compact, high performance, and low cost voip analog terminal adapter (fxs gateway). It comes with one fxs port to interface with a traditional analog phone set or a pbx trunk line for voip communications. By connecting a pstn line to the bypass port, the phone set connected to the fxs port can also access the pstn line for traditional telephone service. The ht-812p is a full featured fxs gateway and is designed for easy installation and configuration. It is an ideal low cost solution for travelers and soho users.
8 GoIP ,GSM VoIP Gateway: The GSM Fixed Wireless Terminal 8 ports Gateway GoIP-8 is a 8 SIM Card Broadband Phone Gateway that had been developed by CHINASKYLINE Ltd. GoIP_8 SIM Card Broadband Phone Gateway is a new product that connect the GSM and the VOIP seamlessly. To GoIP_8 what is installed on the Mobile SIM Card, you can register the GSM telephone on the VoIP soft switch. SIP and H.323 agreement are built in the GoIP_8 and configured flexible. Caller ID can be seen by using SIP. Flexible routing can meet the need of all kinds of call forwarding; even more special is that GoIP_8 support multi-device group, it can be easily combined into arbitrary number of channels of Large Gateway Group. Key Features: Multiple GoIP8 grouping mode Provide one cellular channels for IP-PBX Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2) Single or Multiple Server Registrations Two 10/100 Ethernet circuits connect to the LAN and an additional device GSM module for making GSM calls Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer VLAN and QoS support NAT Transversal and Router functions Voice prompts, HTTP Web, Auto Provision support for configuration and updates Highly stable embedded Linux operating system in high performance ARM 9 Processor Basic Features LEDs for Power, Ready, Status, WAN, PC, GSM Call forward from GSM to VoIP and VoIP to GSM Dial in mode or dial out mode only Dial Plan Password protection for both GSM dial in or dial out Retransmit GSM Caller ID to VoIP terminal Enhanced Features Dynamic selection of codec Advanced jitter buffer Automatic traversal of NAT and firewall VLAN / Qos Router Echo cancellation for Speakerphone Comfort noise generation (CNG) Voice activity detection (VAD) Auto provisioning (requires auto provisioning server) On line firmware upgrade Multi-language support: English and Chinese Hardware Specifications Processor: ARM9E 133MHz DSP: VPDSP101 196MHz Memory: RAM 16MB/ Flash 4MB GSM Module: Type: 850MHz, 900MHz, 1800MHz, 1900MHz Power: Input AC100V ~ 240V, output DC12V/2A +-10% Power consumption: 32W maximum Network card: 100/10Base-T x2 LED: Operation and lines light GSM Passway:eight Operating temperature: 10C to 40C (32F to 104F) Storage temperature: 0C to 50C (32F to 122F) Working Humidity: 40% ~ 90% Not congealed Weight: 1203 g (1 lb) (Including AC/DC Adapter) Warranty: one year. Size: 38*17*7 Weight¡êo1.5Kg
4-channel gsm voip gateway,Goip-4 The 4-channel GSM VoIP Gateway/GoIP-4 bridges the GSM and the IP networks by enabling voice communications. It is ideal for VoIP to Local termination where a fixed telephone line (PSTN) is not available or for cellphone roaming via the a VoIP network. Significant savings on long distance charges can be realized. Key Features : - Open Standard VoIP Protocols (ITU H. 323 V4 and IETF SIP V2) -Single or Multiple Server Registrations -Two 10/100 Ethernet circuits connect to the LAN and an additional device -GSM module for making GSM calls -Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer VLAN and QoS support -NAT Transversal and Router functions -Voice prompts, HTTP Web, Auto Provision support for configuration and updates -Highly stable embedded Linux operating system in high performance ARM 9 Processor Basic Features : -LEDs for Power, Ready, Status, WAN, PC, GSM -Call forward from GSM to VoIP and VoIP to GSM -Dial in mode or dial out mode only -Dial Plan -Password protection for both GSM dial in or dial out -Retransmit GSM Caller ID to VoIP terminal Enhanced Features -Dynamic selection of codec -Advanced jitter buffer Automatic traversal of NAT and firewall -VLAN / Qos -Router -Echo cancellation for Speakerphone -Comfort noise generation (CNG) -Voice activity detection (VAD) -Auto provisioning (requires auto provisioning server) -On line firmware upgrade -Multi-language support: English and Chinese Supported Standards -ITU: H. 323 V4, H. 225, H. 235, H. 245, H. 450 -RFC 1889 - RTP/RTCP -RFC 2327 SDP -RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals -RFC 2976 SIP INFO Method -RFC 3261 SIP -RFC 3264 Offer/Answer model with SDP -RFC 3515 SIP REFER Method -RFC 3842 A Message Summary and Message Waiting Indicator -RFC 3489 Simple Traversal of User Datagram Protocol (UDP) Through Network -AddressTranslators (NATs) -RFC 3891 SIP Replaces Header -RFC 3892 SIP Referred-By Mechanism draft-ietf-sipping-CC-transfer-04 Session Initiation -Protocol Call Control - Transfer -Codec: G. 711 (A Law), G. 729A/B, G. 723.1 -DTMF: RFC 2833, In-band DTMF, SIP INFO Physical and Environmental -Operating temperature: 10 C to 40 C (50 F to 104 F) -Storage temperature: 0 C to 50 C (32 F to 122 F) -Weight: 500 g (1 lb) (Including AC/DC Adapter) -Size: 105 mm (W) x 100 mm (L) x 28 mm (H) (Width x Length x Heigth -Power: 12 Vdc 500 mA (AC/DC adapter included). Siae:36*21*7 Weight:1KG.
Goip_4 gsm gateway is the gsm network and the voip network connecting seamlessly new productss. Mobile phone sim cards will be installed in goip, users of gsm phones can be registered to voip softswitch system. Goip users can be achieved through the gsm network on the car alight. Goip built-in sip and h.323, configuration flexibility. Sip can be thoroughly used when electricity came display numbers. Goip is pbx vendors, system integrators and provider of choice for both gsm productss
If you want to build your own VOIP business ? Contact me ! No more hesitation ! You could setup your own Voip termination system with our best SK Gateway , Goip gateway ,Vos Softswitch , HK/USA server . Our professional tech team will support you day and nights freely .
Fwt ys-ft2008 Fwt ys-ft2008 is a fixed wireless terminal designed to provide better wireless connectivity with high sensitivity to receive signal and large transmitting power to expand extensively the effective coverage of gsm service. It can be used not only in communication for vehicles such as automobiles, trains, floating vehicles, etc., but also in communication for construction sites, isolated islands and rural areas where hardwire-line telephone service is unavailable. So they are widely used in home &office, public phone, rural areas, steamer, railway
. 2 FXS phone ada[pter , Support SIP protocol 2. One RJ-45 port for 10Base-T Ethernet connection 3. Supports Dynamic Host Configuration Protocol (DHCP) 4. Call Waiting,Transfer,Call Forwarding: No Answer / Busy / All 5. Call Return,Call Back on Busy,Call Blocking with Toll Restriction 6. Supports G.711, G.726, G.729, and G.723.1 codec 7. Supports DTMF tone detection and generation 8. Web-based configuration through a built-in web server
JR-900 IP phone is an internet based voice network phone terminal. PHONE series IP phone 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. JR-900 IP phone supports IAX2,SIP,protocols,Support Bridge and Router model.Support IAX2 and dual public server, offers Two Ethernet interface and is compatible with various softswitch systems and VoIP voice gateways to provide broadband IP voice service. Features Support SIP 2.0 (RFC3261) and correlative RFCs Supprt IAX2 Codec:G.711A/u, G.7231 high/low, G.729, G.722 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 speakerphone NAT transverse:support STUN client SIP support SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call SIP support Pubic & Private server. Can connect to Public SIP and Private SIP server at the same time DTMF:Support SIP info, DTMF Relay, RFC2833 SIP application: support Call forward/transfer/holding/waiting Call control features: Flexible dial map, Hotline, Empty calling reject, Black list for reject authenticated call, limit call, No disturb, Caller ID. Support three way conference call Support voice record, 240 seconds max or 3 record max, user-defined record prompt with 1 minute max Support Phonebook 500 records Incoming calls / Outgoing calls / Missing calls. Each support 100 records. Support conference and voice record on SIP server 8 kind of ring type English, Spanish, Czechoslovak alternative Network Features WAN/LAN: Support Bridge and Router model 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 VPN (L2TP/UDP TUNNEL) Support DNS Relay, SNTP Client, Firewall Network tools in telnet server: Including ping, trace route, telnet client Maintenance and Management Web ,telnet and keypad management Management with different account right Upgrade firmware through POST mode Upgrade firmware through HTTP, FTP or TFTP. Telnet remote management./ Upload/download setting file Safe mode provide reliability Supoort Auto Provisioning (upgrade firmware or configuration file) Support Syslog 10PCS/CARTON
VOIP gateways.
