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The Vega 200G VoIP Gateway is the most resilient dual-span gateway in its price class.

The Vega 200G VoIP gateway connects digital telephony equipment to IP networks. Each unit purchased is factory configured with a maximum capacity of 60 simultaneous calls.

The available Sangoma Resiliency Enablement Suite (SRES) makes the Vega 200 the most resilient gateway in it is class. In the event of a WAN failure, IP phones behind the Vega gateway can continue to call each other, be routed to a backup switch or connected directly to the PSTN. In addition, the use of the Sangoma Network Appliance Provisioning (SNAP) tool makes the Vega 200 the easiest to provision gateway on the market.

 

  • Fixed configuration of 60 VoIP calls
  • Supports dual gigabit Ethernet connections
  • EXCLUSIVE Local survivability: Calls on your network stay up in the event the phone network goes down.
  • Configuration is SNAP with the Sangoma Network Appliance Provisioning GUI tool.
  • Flexible call routing for fallback and least cost routing
  • Emergency PSTN backup
  • Voice, fax and modem Support
  • Interoperability with a wide range of legacy and IP equipment

Each E1/T1 interface can be independently configured as network side or terminal side. The Vega 200 gateway can therefore be connected to a PBX or the PSTN. This configuration provides:

  • No disruption to the configuration of existing equipment
  • Flexibility & choice for call routing

مشخصات فنی


VoIP

 SIP

 Fax Support – up to G3 Fax, using T.38

 Modem Support – up to V.90, using G.711

 Up to 60 VoIP channels

  •        Audio codecs:
  •         G.711 (a-law/µ-law) (64 kbps)
  •         G.729a (8 kbps)
  •         G.723.1 (5.3/6.4 kbps)
  •         G.726

 

Telephony Interfaces

  • Primary Rate ISDN (User configurable NT/TE):

E1

  •     Euro–ISDN
  •     ISO QSIG
  •     VN4
  •     CAS R2MFC

 

LAN Interfaces

  •     2 RJ–45s, 1000 BaseT / 100 BaseTX / 10 BaseT, full / half duplex

ویژگی های محصول

 

Identification

  •     Caller ID presentation
  •     Caller ID screening allows connections to be accepted only from selected call sources
  •     SIP Registration & Digest Authentication

 

Operations, Maintenance & Billing

  •     HTTP(S) web server
  •     RADIUS Accounting & Login
  •     Remote firmware upgrade:
  •         Auto code upgrade
  •         Auto configuration upgrade
  •     SNMP V1, V2 & V3
  •     TFTP/FTP support
  •     VT100 – RS232

Routing & Numbering

  •     Dial Planner – sophisticated call routing capabilities, standalone or gatekeeper/proxy integration
  •     Direct Dialing In (DDI)
  •     SIP registration to multiple proxies
  •     NAT traversal

Security & Encryption

  •     Management – HTTPS, SSH, Telnet
  •     Configurable user login passwords

 

Call Quality

  •     Adaptive jitter removal
  •     Comfort noise generation
  •     Silence suppression
  •     802.1p/Q VLAN tagging
  •     Differentiated Services (DiffServ)
  •     Type of Service (ToS)
  •     QoS statistics reporting
  •     Echo cancellation (G.168 up to 128ms)

مستندات

منبع


احساس رضایت را با ساعیان ارتباط آینده پیشرو تجربه کنید
آدرس : تهــــران، خیابان مطهـــــری
خیابان اورامان، پلاک ۳۴، واحد ۱۰۴
تلفن‌:‌
۰۲۱   88315442
۰۲۱   ۸۸۳۱۶۷۰۴
۰۲۱   ۸۸۳۱۶۷۱۸
۰۲۱   ۸۸۸۲۶۱۱۷
۰۲۱   ۸۸۳۱۵۳۸۴
[email protected]
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