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VoIP Gateway

VoIP Gateway mempunyai 2 fungsi :

- Sebagai FXO dan FXS Gateway : Sebagai interface agar terkoneksi ke telkom line ( FXO )  dan 

  sebagai penyedia analog Extension ( FXS )

- Sebagai Peer to Peer Connection agar PABX Panasonic antar kota atau negara dapat menelepon secara gratis via internet

Voice Gateway 4FXO is 4-Line FXO gateway with SIP pr...

VoIP Gateway mempunyai 2 fungsi :

- Sebagai FXO dan FXS Gateway : Sebagai interface agar terkoneksi ke telkom line ( FXO )  dan 

  sebagai penyedia analog Extension ( FXS )

- Sebagai Peer to Peer Connection agar PABX Panasonic antar kota atau negara dapat menelepon secara gratis via internet

Voice Gateway 4FXO is 4-Line FXO gateway with SIP protocol IP
device which allows to connect 4 Lines of analog PSTN telephone
line or connect to analog extension of PABX to make or receive
VoIP call over Internet or VPN network. This device is suitable
for office IP-PBX application at office to office or office to branch
office to call between PSTN Line and IP Call.
To select up to 4 SIP service Accounts


Voice Gateway 4FXO is appropriate to use four VoIP SIP Trunk or IP Centrex
service or IP-PBX within offices and remote branch offices. One of four SIP
Servers ( or ITSP Service provider or alternative IP-PBX ) can be configured
freely at each line ( FXO port ) to make or receive IP Call. It provides 4 service
platforms according to your dial number or routes plan.
IPv6 VoIP Gateway is ready to Market
IPv6 address was developed for years, however, it was not practical to our life
up to date. More and more electronic devices are able to link to IP Network, this
makes existing IPv4 address supply in shortage to global market. Meanwhile,
the emerging countries are not able to increase IPv4 address supply due to
strong market demand on broadband services. Voice Gateway 4FXO is an SIP
based FXO gateway which built-in both IPv6 and IPv4 IP address. No matter
when you are ready to deploy IPv6 network now, or reserve the future
expansion to IPv6 from existing IPv4 address, Voice Gateway 4FXO is ready to
grow up with you. Both IPv6 and IPv4 address are working simultaneously at
Voice IP Call. Its flexibility of both IPv6 and IPv4 accept and interwork both
addresses on today and tomorrow whenever you need.


Flexible Dial plan and Route Plan Features
Voice Gateway 4FXO provides flexible Dial Plan between FXO and IP Trunk
(SIP Soft Switch). Dial Plan is to configure in what condition the digits can be
sent out to/from IP network. The dialing inter-digit time before dialing is
configurable to meet local PSTN line or PBX’s extension line. Dial Rule is able
to detect the prefix code and maximum digits reached and then dial
automatically.


The Digit Manipulation (DM) allows you to configure matched prefix code, digits
length, start and stop digit position to be replaced digits as well.
Route Plan is to configure the incoming and outgoing call routes which you
desired this call to go out or allow to income. For instance, IP incoming call may
Reach to one FXO port with Priority or Cyclic access. You can also configure IP
incoming call by Matched prefix digits, Matched dialing number to FXO line and
Matched digit length. For FXO outgoing call to IP routes, the hunting type
supports Priority or Cyclic or Simultaneously to select which SIP trunk ( SIP
Proxy Server ) to go. FXO outgoing call routes also support by Matched prefix
digits, Matched outgoing SIP Trunk number and Matched digit length. Both
direction supports No Answer time out and Backup Routes.
Suit to IP-PBX to access local PSTN line


Voice Gateway 4FXO is a SIP IP device to connect with IP-PBX to access local
PSTN network with FXO interface. Its telephony features, for instance, Caller ID
detection and Releasing FXO port after call was dropped, are easy to integrate
with Legend Telephony Line with IP-PBX in office and branch office IP call
application. It is compatible with local Telecom network regulation and your
office IP network to transmit analog voice between them.


Specification
· Interface:
o Ethernet port (RJ-45, 10/100 base-T)
1-WAN port, connect to IP Network
1-LAN port connect to PC with NAT
o Support Bridge, NAT and Gateway mode
o Telephony port connect to local PSTN line (RJ-11 x 4 pcs)
o DC +12V power input Jack
o Reset key to return Factory setting
o LED Indicator for System, SIP and FXO status
· IP Network connection
o IPv4 (RFC 791) and IPv6 Simultaneously
o IPv6 Auto Configuration (RFC 4862)
o IPv6 Only, IPv4 Only or dual stack
o MAC Address (IEEE 802.3)
o MAC Clone Setting
o Vendor Class ID
o IP/ICMP/ARP/RARP/SNTP
o Static IP
o DHCP Client (RFC 2131), WAN port
o DHCP Server, LAN port
o NAT Server (RFC 1631)
o PPPoE Client
o DDNS ( DynDNS )
o DNS Client
o Firewall
o URL Filter
o IP Filter
o MAC Address Filter
o Application program Filter
o Port Filter
o Port Forwarding (TCP, UDP or both)
o Bandwidth Control (Download and Upload), Maximum Bandwidth
priority setting
o UPnP Server at LAN port
o Behind NAT, use DMZ for NAT traversal
o SNTP with time zone and Daylight Saving
o TCP/UDP (RFC 793/768)
o RTP/RTCP (RFC 1889/1890)
o IPV4 ICMP (RFC 792),
o TFTP Client
o VoIP VLAN Support 802.1Q, 802.1P
o VLAN ID Range : 2 to 4094
o VLAN Priority : 0 to 7 (Highest Priority)
o QoS : DiffServ (RFC 2475), TOS (RFC791, 1394)


· SIP Protocol :
o RFC3261 compliance
o Support up-to 4 SIP Trunk to Register
o SIP UDP Protocol
o Support SIP compact Form
o Support SIP HOLD Type: Send Only, 0.0.0.0 or inactive
o SIP Session Timer (RFC 4028)
o SIP Session Refresher: UAC or UAS
o SIP Encryption
o MD5 Digest Authentication (RFC2069/RFC2617)
o Reliability of provision response PRACK (RFC3262)
o Early/Delay Media support
o Offer/Answer (RFC3264)
o Message Waiting Indication (RFC3842)
o Event Notification (RFC3265)
o REFER (RFC3515)
o Support Outbound Proxy
o Support Primary and Backup SIP Server
o Support STUN NAT Traversal
o Support “rport” parameter (RFC 3581)
o Configure SIP local Port
o SIP QoS Type: DiffServe or QoS
o Accept Proxy Only : YES or NO


· Audio Codec :
o G.711 A-law/μ-law, G.729A, G.723.1 (6.3K, 5.3K)
o Select voice codec priority : Local or Remote
o Voice Payload size (ms) configuration
o Silence Suppression
o VAD/CNG
o LEC : Line Echo Canceller
o Max Echo Tail Length (G.168): 32, 64 and 128ms
o Packet Loss Compensation
o Automatic Gain Control
o In-band/out of band DTMF (RFC4733, RFC2833 / SIP INFO)
o Adaptive/Configurable Jitter Buffer
o G.168 Acoustic Echo Cancellation
o Configure RTP basic Port
o RTP QoS Type : DiffServ or TOS
o Phone Book ( 50 records ) for peer to peer calls
o Dialing Plan with drop, replace, Insert dialing digits
o Select First digit and Inter digit timeout duration (Sec)
o Selectable Call Progress Tone
o Support Specified Line Calling


· Call Features :
o 4-Line FXO connect to PSTN or PBX simultaneously
o Caller ID recognition DTMF (before/after 1st ring) and FSK (before
1st ring ), ETSI
and Bellcore
o DTMF Caller ID start and stop BIT configurable
o Current Drop Detection to release FXO port
o Disconnect tone recognition to release FXO port
o Tone Generation: Ring Back, Dial, Busy, call waiting, ROH,
Warning, Holding, Stutter
dial tone and disconnect tone
o Configure Tone Frequency, Cadence, Level and Cycle
o Select Tone specification by Country name List
o Global Country Based Tone Specification
o NAT Traversal support STUN, UPNP and Behind NAT
o Out-Band DTMF : RFC2833 and SIP Info
o RFC2833 Payload type : 101 or 96
o DTMF send out ON and OFF Time configure
o DTMF incoming recognition Minimum ON and OFF time
o DTMF Relay Volume configuration
o T.38 FAX Volume configuration
o Flash Time transmit via SIP Info (Enable or Disable)
o Message Waiting Indication (Stutter Tone Notice)
o Block Anonymous Call
o Call Hold
o Call Transfer


· FXO Line Configuration:
o Activate or deactivate
o Line ID
o FXO Line Phone number
o Polarity Reversal detection for call establish and Billing
o Current drop recognition to release port
o Incoming call Handle: Hotline or 2 stage dialing
o HOT Line to desired phone number
o Play voice file to incoming call
o Repeat playing voice file counts
o Self-recorded voice files to upload
o Generate FLASH TIME to PSTN network
o T.38 or FAX Relay Type
o Incoming and outgoing dB value configurable
o Dialing Answer Delay time to establish call path
o Answer PSTN incoming call after how many ring cycles
o Caller ID detection mode by Country selection
o VoIP dial to FXO/PSTN Line: 1 stage dialing and 2 stage dialing
o Outgoing SIP Caller ID Selection
o Support 4 SIP Trunk
o Accept desired SIP Proxy incoming calls Only


· Flexible Routing Plan :
o Prefix Match and Length
o Priority Ring
o Cyclic Ring
o Simultaneous Ring
o Programmable Hunting Cycle
o Backup Routes with Digit Manipulation
o Default Routes


· Flexible Dial Plans :
o Retrieve transfer call from 3rd party by dial Code (default: *#)
o Inter digit time out setting
o First digit dial out delay time setting
o End of dial keypad number
o Dial Rule : Match dial Prefix and Maximum digits length ( 1-15 )
o Phone Book can be Exported or Imported
· Digit Manipulation (Drop and Replace Rule):
o FXO DM Group
o VoIP DM Group
o DM 1 Group
o DM 2 Group
o DM 3 Group
o DM 4 Group
o Matched Prefix
o Matched digit length
o Replace digit start position
o Replace digit stop position
o Replace number
o Incoming Ring frequency recognition range: 10 to 70 Hz
o Incoming Ring ON time recognition range: 0 to 8000ms
o Incoming Ring OFF time recognition range: 0 to 8000ms
o Incoming Ring Level recognition range: 10 to 95Vrms
o Support Peer to Peer Dialing
o Flash Time Detection: range from 80 to 800 ms
o Configure Ring Cadence, Frequency and Voltage


· MANAGEMENT :
o Administrative Telnet CLI and HTTP, HTTPS
o HTTP provision through MAC address
o Multilingual Web User Interface
o 3 Levels of User Access Right with Password protection with
different Web Language
(Administrator, Supervisor and User)
o HTTP/HTTPS Service Access limitation from WAN port
o Configure Service ports at HTTP, HTTPS and telnet Services
o Phone Debug Module: Device Control, Call Control, DB, Verbose
o SIP Debug Module: Register, Call, SIP Message, Others
o SNTP Debug Module
o Device Debug Module
o DSP Debug
o Provide 8 Debug Levels :
Emergency
Alert
Critical
Error
Warning
Notice
Information
Debug
o Provides System Status Logs
o Connect to external SYSLOG Server
o Status display: Network, Line, SIP Trunk status
o Diagnostics (debug through Syslog Event Notice)
o Debug in real time by Telnet
o Auto Provision via HTTP Server
o SNMP V2/Trap
o Configuration Backup/Restore
o Dual Firmware Image Backup
o Reset to factory Default
** Support Welltech proprietary encryption protocol at SIP Signal and
Voice codec during
transmitting to IP network in order to Anti-ISP block of VoIP call. This
feature only be
available with Welltech SIP server or SIPPBX6200 IP-PBX


· Environmental :
o Actual Dimension: 17.5(W) × 3.2(H) × 12.6(D) CM
o Weight: 0.5kg (One unit with packing)
o Operating Temp. & Humidity
Temp.: 0°C~45°C (32°F~113°F)
Humidity: 10%~90% relative humidity, non-condensing
o Power Adaptor:
INPUT: AC100V~240V, 50/60Hz
OUTPUT: DC 12V, 1.5A
· Approvals:
o CE, FCC (Part 15, Class B), LVD and RoHS
· Country of origin:
o Made in Taiwan
· Packing Accessories
o Voice Gateway 4FXO x 1 pcs
o AC to DC+12V Power adaptor x 1 pcs
o CD User Manual x 1 pcs
· Warranty
o One year

SEGERA BELI DI SOPHOS TECHNOLOGY

cs@sophos-technology.com

021-2907-3411,  021-504-33777

More

VoIP Gateway There are 27 products.

Subcategories

  • SATURNUS VOIP GATEWAY

    Saturnus Voip gateway dengan features fitur  Auto Learn Disconnect Tone untuk integrasi ke PABX Panasonic analog konvensional.  Harga murah Jakarta

  • WELLGATE voip gateway

    WELLGATE VOIP GATEWAY berfungsi untuk mengintegrasikan 2 buah PABX Panasonic agar bisa telepon GRATIS antar kota / negara

    sudah terkenal di Indonesia karena kualitas premium, tahan lama dan jarang rusak.  Suara jernih.  

  • OPENVOX voip gateway

    VOXSTACK VOIP GATEWAY Tersedia tipe FXO dan FXS, isi 8 port sampai 44 ports, berfungsi MODULARM

    dimana Module FXO dan FXS dapat dicabut keluar.  Jika module rusak tinggal ditarik keluar dan diganti dengan module yang baru, sehingga  tidak mengganggu Operasional

    Untuk tipe iAG - VoIP Gateway bersifat stand alone, tersedia 4 fan 8 port FXO atau FXS

  • PLANET voip gateway

    Planet VOIP gateway isi 4 port, 8 port, 16 port, 24 port, 32 port,  tipe FXO atau FXS atau kombinasi FXO + FXS

  • GRANDSTREAM voip gateway
    • The GXW series includes two models with 4 or 8 ports respectively
    • Designed and tested for full interoperability with leading IP-PBXs, soft-switches and SIP-based environments
    • Manageability, a simple configuration, superb voice and video quality and feature rich functionality
    • Based on open industry standards
  • Quintum atau Sonus

    • 2, 4, 6 or 8 Analog line and trunk interfaces
    • Up to 8 simultaneous VoIP calls
    • Available in MultiPath or Gateway configurations
    • Auto-Provisionable

  • NETPHONIC VOIP GATEWAY

    NG-48 series of gateways are innovative gateway that offer a rich set of
    functionality and superb sound quality. They are fully compatible with
    SIP and H.323 industry standard and can interoperate with many other
    SIP or H.323 compliant devices and software on the market.

  • ATA
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Showing 1 - 16 of 27 items
  • $0

    WellGate 3512 is a Wi-Fi VoIP Gateway, which built-in 2-port FXS VoIP ATA, 1-WAN/4-LAN router IP Gateway, and 802.11 b/g WLAN together. It supports SIPv2 (RFC3261) protocol and with rich telephony features. Wireless LAN can be configured to either Access Point (WiFi AP) or Client mode, with automatically site scan and access control by MAC address. It...

    $0
  • $1,200

    OpenVox VoxStack Series Analog Gateway is an industry 1st open source asterisk-based Analog VoIP Gateway solution for SMBs and SOHOs. With friendly GUI and unique modular design, users may easily setup their customized Gateway. Also secondary development can be completed through AMI.

    $1,200
  • $310

    The GXW series includes two models with 4 ports respectively Designed and tested for full interoperability with leading IP-PBXs, soft-switches and SIP-based environments Manageability, a simple configuration, superb voice and video quality and feature rich functionality Based on open industry standards

    $310
  • $0

    • 2, 4, 6 or 8 Analog line and trunk interfaces • Up to 8 simultaneous VoIP calls • Available in MultiPath or Gateway configurations • Auto-Provisionable

    $0
  • $0

    NG-48 series of gateways are innovative gateway that offer a rich set of functionality and superb sound quality. They are fully compatible with SIP and H.323 industry standard and can interoperate with many other SIP or H.323 compliant devices and software on the market.

    $0
  • $0

    PROMO USD 233 ( 8FXS ) USD 359 ( 8FXO ) - HANYA  SEPT-OKT-NOV-2016. Berfungsi untuk menyambungkan PABX Panasonic agar bisa telp GRATIS 24 jam, antar Kota / negara. Kualitas Grade-A, spare part Taiwan, jarang rusak, tahan lama, suara jernih, tidak putus-putus Juga berfungsi sebagai FXO Extension analog

    $0
  • $0

    Berfungsi menyambungkan PABX Panasonic agar bisa telp GRATIS 24 jam, antar Kota / negara Kualitas Grade-A, spare part Taiwan, jarang rusak, tahan lama, suara jernih, tidak putus-putus.   Juga berfungsi sebagai FXS Extension analog Segera beli di Sophos Technology

    $0
  • Berfungsi untuk menyambungkan PABX Panasonic agar bisa telp GRATIS 24 jam, antar Kota / negara Kualitas Grade-A, spare part Taiwan, jarang rusak, tahan lama, suara jernih, tidak putus-putus.   Juga berfungsi sebagai FXO Extension analog

    $0
  • $0

    Berfungsi untuk menyambungkan PABX Panasonic agar bisa telp GRATIS 24 jam, antar Kota / negara. Kualitas Grade-A, spare part Taiwan, jarang rusak, tahan lama, suara jernih, tidak putus-putus Juga berfungsi sebagai FXO Extension analog

    $0
  • $0

    Berfungsi untuk menyambungkan PABX Panasonic agar bisa telp GRATIS 24 jam, antar Kota / negara. Kualitas Grade-A, spare part Taiwan, jarang rusak, tahan lama, suara jernih, tidak putus-putus Juga berfungsi sebagai FXO Extension analog

    $0
  • $0

    - Auto Learning Disconnect Tone - memudahkan untuk integrasi ke PABX Panasonic dan telkom Line, langsung berfungsi.  Menjamin Line Panasonic & PSTN tidak hang / nyangkut.  Auto release. HARGA SAAT INI USD 260  ( 4 FXO )ADALAH HARGA PROMO HARGA NAIK 31 DESEMBER 2015

    $0
  • $1,300

    OpenVox VoxStack Series Analog Gateway is an industry 1st open source asterisk-based Analog VoIP Gateway solution for SMBs and SOHOs. With friendly GUI and unique modular design, users may easily setup their customized Gateway. Also secondary development can be completed through AMI.

    $1,300
  • $740

    Voxstack-24FXS berisi 24 port FXS, dimana FXS berfungsi sebagai extension analog Voxstack termasuk dalam produk OPENVOX yang terkenal dengan kualitas premium, tahan lama, harga terjangkau. Garansi 1 tahun, dijamin Hubungi kami 021-2907-3411

    $740
  • $1,200

    Voxsatck  40FXS berisi 40 port FXS, fungsi FXS sebagai extension analog. Voxstack 40FXS termasuk merek OpenVox, dan OpenVox sudah terkenal dengan kualitas bagus,  harga terjangkau Segera beli di Sophos Technology

    $1,200
  • $0

    VoIP Gateway murah isi 4FXS hanya USD  155, termurah.  It can be configured for different country uses, provides a wide selection of codecs including G.711A、G.711U、G.729、G.722、G.723、ILBC.The iAG Analog VoIP Gateways use standard SIP protocol and fully compatible with Leading IMS/NGN platform, IPPBX and SIP servers. 

    $0
  • $0

    VOIP Gateway termurah 8 FXO merek OpenVox, dilengkapi fitur AUTO LEARN DISCONNECT TONE,  cocok untuk integrasi Peer to Peer 2 buah PABX Konvensional Panasonic supaya bisa telp gratis antar cabang, antar kota Beli segera di Sophos Technology 021-2907-3411,   cs@sophos-technology.com

    $0
Showing 1 - 16 of 27 items