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Positron G-600V, IP PBX Video Conferencing System, 4 x ETH, 4 X T1/E1 PRI, 2 x FXO, 2 x FXS, up to 250 Users

$0.00

Description

he G-600V video conferencing feature is an integral part of the G-600 feature rich IP PBX. The G-600V offers a complete video solution enabling video-capable endpoints to establish video calls and participate in a video conferencing bridge without extra software license costs. The G-600V video conferencing improves interaction between people, productivity and efficiency. It is intended for desktop video conferencing on SIP enabled video phones and on PC’s or tablets with soft phones. Every position/user or remote caller, if equipped with a videophone or soft phone, whether local or remote, can participate in a video conference. A video conference can be set up as easily as setting up an audio conference call.

The G-600 IP PBX is the ideal solution for medium enterprises that can accommodate up to 250 users. It enables a powerful suite of communication applications to launch enterprise communication into the next era and that completely replaces legacy proprietary phone systems, supports standard SIP soft/hard phones from any vendor, VoIP service providers and integrates the telephony network interface.

The base of the G-600 is a feature rich IP PBX business phone system that will increase the productivity of employees, reduce the support time and operate at a much lower cost than traditional phone systems. To increase productivity, the system will transparently allow users to receive calls while they are in the office, on the road or working from home. The G-600 IP PBX is a standalone solution that can be installed on any network quickly and easily. For IT staff, the system supports an easy to use web interface that supports wizard based setups.

Additional information

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  • Installs simply and quickly
  • 4 T1/E1 PRI
  • 2 Analog Phone Lines (2 FXO)
  • 2 Analog extensions (2 FXS) for Fax or Handsets
  • Supports VoIP out of the box ­ no upgrades required

Software Features  
Audio Conferencing :

  • Meet-me style room   
  • Flexible management of number of conferences   
  • Participant code access   
  • Moderator lock   
  • Music-on-hold per conference

 
Caller ID

  • Caller ID for outgoing calls on SIP lines • 
  • Caller ID for internal calls and remote offices   
  • Caller ID blocking   
  • Caller ID on call waiting   
  • Caller ID prefix and suffix   
  • Caller ID per user, per area code, global

Call Queues

  • Agent log in  
  • Skill-based routing  
  •  Priority queuing  
  •  In queue music or messaging  
  •  Remote queue membership   
  • Queue announcements   
  • Queue visualization   
  • Caller breakout options

 
Intercom and Paging

  • Zone paging  
  •  Desktop paging  
  • Extension paging 
  • Overhead paging

Phone Provisioning

  • Automatic phone provisioning using:
    •  DHCP    
    • SIP SUBSCRIBE option (Yealink, Snom)    
    • Multicast DNS option (Aastra) 
    • Manual phone provisioning option  
    • Interoperability tested with Yealink, Snom  Aastra, Polycom and Cisco SPA brands
    • Expansion module provisioning   (Yealink, Snom, Aastra) •
    • Custom phone templates

User Provisioning

  • User bulk provisioning option 
  • User template assignment  
  • Mobile phone dispatch option  
  • Courtesy phone extension 
  • Forward external number only (secretary call screening)

Calling Features

  • Call park 
  • Call hold  
  • Park and page 
  • Ring groups (ring all, in sequence, audio-out)  
  • Playback option per ring group  
  • Follow me with caller announcement  option  
  • Dial by name directory 
  • Phone number alias 
  • Group call pickup
  • Directed call pickup  
  • Spy call monitoring  
  • Direct inward dialing  
  • Call routing per schedule  
  • Call routing per IVR selection  
  • Call routing per caller ID  
  • Call forward all • Attended transfer  
  • Blind transfer • Busy Lamp Field  
  • On-demand call recording –1 call at a time  
  • Strict call accounting 
  • Flexible outgoing call routes  
  • Failover outgoing call routes  
  • Flexible incoming call routes  
  • Remote users • Inter-branch calling  
  • Point-to-point video calling  
  • Incoming URI calling 
  • Audio-in extension  
  • Routing to user voice mailbox  
  • Corporate call back (DISA)
  • 15 concurrent call channels

Voicemail 

  • Deposited on plug-in USB memory stick  
  • Custom voicemail greetings  
  • Voicemail-to-email 
  • Voicemail forwarding  
  • Folders per mailbox 
  • Voicemail management from user portal
  • Caller breakout from voicemail options

Automated Attendant 

  • Unlimited steps  
  • Cascaded IVRs  
  • On-demand time frames 
  • Time-day-date service  
  • Voice prompts  
  • Music on hold (GSM files, audio-in)  
  • User authentication  
  • Custom sound manager  

 System Management

  • Web-based (http) administrator interface 
  • Remote management capability     
  • Web-based user portal  
  • Operator console for call handling  
  • Backup and restore configuration 
  • PBX and network diagnostics  
  • Call recording management 
  • Call detail records management

Hardware Specifications Features

  • Single Compact Flash port 
  • Integrated T1/E1, FXS and Ethernet  
  • 4 x T1/E1 PRI for connection to the telephone network  
  • Two Foreign Exchange Station (2 FXS)  and two FXO ports for connecting a phone or fax machine  
  • Four Ethernet ports (3 LAN, 1 WAN) 10/100 Mbps auto MDIX  
  • Hardware based G.168 echo cancellation chip  
  • Rack mountable

Interfaces 

  • Telephone interface:
    • 4 x T1/E1 RJ-45,
    • 2 x FXS RJ-11
    • 2 x FXO RJ-11 
  • Network interface: four 10/100 Mbps RJ-45  
  • LED indicators: Power, Fault, Network, Link Speed  
  • Standard audio input: Male mono jack 1/8 
  • Standard audio output: Male mono jack 1/8  
  • USB: USB 2.0 (11 Mbps)

Security

  • Blacklist per IP range 
  • SIP intrusion detection with automatic blocking  
  • Embedded firewall  
  • Access control list for SIP registration 
  • TLS/SRTP for signaling and media

SIP Interface

  • Version 2.0  
  • SIP TLS  
  • P-asserted indentity header  
  • Support for a range of source IP on a single trunk  
  • VoIP provider templates  
  • NAT traversal for remote extensions 
  • DTMF modes: RFC2833, SIP info, inband, auto

Codec Support

  • G.711 u-law & a-law

Dimensions •

  • Width: 19 inches (48.3 cm) 
  • Depth: 20 inches (50.83 cm) 
  • Height: 1.75 inches (4.4 cm)

Environmental •

  • Operating temperature: 32 to 104°F (0 to 40°C) 
  • Storage temperature: -4 to 185°F (-20 to 85°C) 
  • Humidity: 10% – 80% non-condensing

Standards and Approvals 

  • EMC: FCC Part 15 
  • Safety: UL 60950

Warranty •

  • 1 year Ordering Information