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Sipdex 4-8 Port FXO/S Gateways

sipdex voip gateway SIP 4 FXO
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Key Note

  • Supports SIP/MGCP protocols
  • Flexible configurations of FXS/FXO ports
  • Outstanding security: Configurable SIP ports and IP address whitelist
  • Supports IP/PSTN LCR selection
  • Supports fax over IP using T.38
  • Supports polarity-reversal/busy-tone detection
  • Supports Auto Provisioning and TR069/SNMP
  • Supports HA and PSTN failover
  • Supports Class I lightning protection
  • Compatible with standard SIP platforms, such as 3CX, Asterisk, Elastix

 

Cost-effective VoIP trunk gateway

Sipdex 4-8FXO/S Gateway powerful hardware platform provides sufficient power for IP packetization, voice compression and echo cancellation even when the voice traffic is at the peak. Sipdex 4-8FXO/S Ports Gateway is based on the embedded Linux operating system. With its powerful hardware platform and flexible software design, it can meet various voice application requirements.

VoIP

  • Support SIP 2.0 (RFC3261)
  • Redundancy sip server capable
  • Hotline
  • Call Forward、Call transfer、Call hold、Call waiting
  • 3-way Talking、Pickup、Join call、Redial、Unredial
  • Call Park、vport、click to dial
  • DND(Do Not Disturb)
  • Black List,Limit List
  • E.164 dial plan and customized dial rules

FXS

  • Polarity inverse generation
  • Hook flash timing setting
  • Message waiting indicator (FSK, polarity inverse)
  • Caller ID generation (FSK, DTMF, before ring and after ring)
  • Ring cadence setting
  • Ring frequency setting
  • Volume control

FXO

  • WAN/LAN: Support Bridge
  • Polarity inverse detection
  • Busyout
  • Direct inward dialing
  • Automatic attendant (2nd – stage dialing)
  • Ring timing setting
  • Volume control
  • DTMF out-pulsing timing setting
  • Caller ID detection (FSK, DTMF, before ring and after ring)
    Busy tone detection

Codec/FAX/RTP

  • Web ,telnet and keypad management
  • G.711ALaw, G.711ULaw, G.729A, G.723.1r63, G.723.1r53, GSM
  • T.38 fax relay, T.30 fax transparent
  • Echo cancellation
  • Dynamic jitter buffer management
  • Static jitter buffer
  • DTMF relay (RFC 2833, SIP/INFO, inband)

Voice QoS

  • IEEE 802.1p tag
  • DiffServ code point (TOS) bits

Call control

  • Blind/Explicit transfer
  • Call forward on busy/on no answer/variable
  • Call waiting
  • PSTN failover
  • Hunt group
  • Call number transformation (Add, delete, replace)
  • Calling and called number based routing
  • Three-way calling
  • Call progress tone (Configurable)
  • Color ring back tone
  • Music on hold
  • Do not disturb

Security

  • IP access list (IP table)
  • SIP/RTP/TELNET/HTTP/TFTP port assignment
  • Web-utility access privilege (Admin and user)

Networking

  • DHCP
  • DNS/DDNS
  • NAT traversal (STUN)
  • PPPoE

System management

  • TR069-based management (Include TR069, TR104, and TR106)
  • Web-based management interface (Local and remote access)
  • Firmware upgrade
  • Log management (8 levels)
  • Various debugging and call trace
  • Remote management with console and telnet
  • Configuration files import and export
  • System status monitoring and statistics

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Q:How many FXO/S Ports are built in in This gatway.

A:Our gateway could have up to 8 FXS/O Ports

Q:Can I fax through an Your Sipdex Gateway?

A:Yes you can! Our Sipdex Gateway could support T.38, which could make you fax on IP Networking.

Q: Is your Sipdex Gateway compatible with asterisk and related open source Platform?

A:Yes, our Sipdex Gateway could work with opensource pbx software Asterisk, Free PBX, Elastix And Any other IP PBX based on Asterisk.

Q:What voice compression do VoIP gateways support?

A:Our Sipdex Gateways will support standard codecs such as G.711, G.729A, G.723.1,

Q:Your FXO Ports of Sipdex gateway support Polarity Reverse Detetion, Call ID Detection, Echo Cancellation and Second Dial Tone?

A:Yes. Our Sipdex Gateway could support them perfectly.

Q:What types of applications are VoIP gateways used for?

A:Dozens. The three most common applications include:PSTN connectivity for your VoIP phone system Connecting traditional telephones to your VoIP phone system VoIP connectivity for your traditional PBX system

Q:Do your gateway support Routing?

A:Yes absolutely. That’s our super star feature of our Gateway, you could create a flexible routing rule to control the call.

Q: Do your gateway have any tool to debug?

A:Yes, our gateway could have WEB Page to show the status of the gateway. More over, our gateway could output the sip logs,which could help you to know how sip function in our gateway.

Q:I try to config your Sipdex gateway with elastix pick up function, but fail. Any ideas?

A: Please add *8 value in Web config of gateway>>routing>>>Digit Map

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Matrix is VOIP , Asterisk & Elastix Professional Consultancy Company. We based in Hong Kong and Shenzhen,China.

We are able to make all your branches to be connected, it is your own One Communication Platform that is you desire

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