The G-1212 IP PBX is the ideal small business solution for up to 60 users, which combines full function support of a PBX with the integration of both analog and VoIP telephony. It not only supports the telephony needs of small businesses but also offers compelling features such as: “Never miss a call” - where the system will track a person based on the phone numbers provided; “Automated Attendant” - look more professional by having an automated system answer calls and direct users; “Conference Rooms” - never pay for multiple person conference bridges again, which saves money. The business phone system has four Ethernet ports, two Foreign Exchange ports for connecting to a phone or fax machine, an echo canceller and functionality for connecting telephone lines and two USB ports, transforming this into a truly powerful and flexible system. The G-1212 IP PBX is a standalone solution that can be installed on any network quickly and easily with the aid of its web browser based configuration that supports wizard based setups.
- Installs simply and quickly
- 12 Analog Phone Lines (12 FXO)
- 2 Analog extensions (2 FXS) for Fax or Handsets
- Supports VoIP out of the box no upgrades required
- Four Foreign Exchange Office (4 FXO) ports for connection to the telephone network
- Two Foreign Exchange Station (2 FXS) ports for connecting a phone or fax machine
- Single Compact Flash port
- Two USB ports
- Audio input / output jacks
- Hardware based G.168 echo cancellation chip
- Four Ethernet ports (3 LAN, 1 WAN) 10/100 Mbps auto MDIX
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
Trunk Types •
- SIP •
- Analog •
- Music-on-hold per trunk
Hardware Specifications
- 12 FXO RJ-11
- 2 FXS RJ-11
- 4 Gigabit Ethernet RJ-45
- Standard audio input mono jack 1/8
- Standard audio output mono jack 1/8
- Hardware based G.168 echo cancellation chip
- Rack mountable
- Real-time clock with independent power source
- Per-country tone indications
LAN Interfaces •
- 1 WAN Ethernet Port
- 3 LAN Ethernet Ports
Codecs
- G.711 u-law & a-law
- H.264 Network Services
- DHCP client and server
- NTP client and server
- TFTP server
- LAN and WAN monitoring
Security
- Blacklist per IP range
- SIP intrusion detection with automatic blocking
- Embedded firewall
- Access control list for SIP registration
SIP Interface
- Version 2.0
- 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
Storage
- USB interface memory stick Dimensions
- Width: 12 inches (30.5 cm)
- Depth: 5.25 inches (13.3 cm)
- Height: 1.83 inches (4.6 cm)
Environmental
- Operating temperature:
- Storage temperature:
- -4 to 185°F (-20 to 85°C)
- Humidity: 10% - 80% non-condensing
Warranty
- 1 year Ordering Information • Product code: 70-00058
