Bellus SIP-1112 VoIP Terminal
The Inter-Tel Bellus SIP phone series offers a complete lineup of user-friendly VoIP phones that are also eco-friendly. Through its intuitive and pleasing interface, our SIP terminals deliver fully-featured, cost-effective voice and data communications. Inter-Tel defines the standard for low ownership costs through our energy-saving technologies, and we help preserve our precious resources through the use of recyclable and reground plastics.
Bellus SIP Phone SIP-1112 Feature Set
- Support for up to 6 VoIP accounts on the same terminal
- National language selection
- An anti-glare, white backlit LCD
- Full-duplex high-definition speakerphone
- Call history: dialed, received, missed and forwarded calls
- Local phonebook up to 1000 entries
- Call Waiting
- Call Forwarding
- Call Transfer
- Three-way conferencing
- SMS support
- Black list
- One-touch speed dial
- Power over Ethernet (IEEE 802.3af), Class 2
- Headset and EHS support
- Supports expansion modules
- Simple, flexible and secure provisioning options
* Support for these features may vary with your service provider
Configuration Management
Settings on the Bellus SIP-1112 are managed through the phone, a standard web browser, or auto-provisioning. Separate user and administration modes, as well as a user keypad lock for personal privacy protection are standard with Bellus SIP phones.
Security protocols:
- Support for HTTPS Server and Client, TLS (end-to-end), digest authentication using MD5/MD5-sess, AES encrypted data packet transfer, and SRTP are included
- HTTPS certificate manager
- OpenVPN
- IEEE 802.1X
- IPv6
Network management features:
- Static, DHCP or PPPoE IP address assignment
- FTP/TFTP/HTTP/HTTPS auto-provisioning for mass deployment
- Auto-provisioning with PnP
- STUN (Simple Traversal of UDP through NATs) mode is supported
- QoS: 802.1p/Q tagging (VLAN), Layer 3 ToS/DSCP
- Time and date synchronization using SNTP
- UDP/TCP/DNS-SRV (RFC 3263)
- Call server redundancy supported
- Proxy mode and peer-to-peer SIP link mode
SIP features supported
- VoIP calls can be initiated with or without a SIP proxy
- Fully compliant with RFC 2543 and RFC 3261
- Call server redundancy supported
- Proxy mode and peer-to-peer SIP link mode
Audio codecs supported
- Wideband (HD): G.722
- Standard codecs: G.711 / G.723.1 / G.726 / G.729ab
- Additional codec: iLBC
DTMF features:
- In-band, Out-of-band (RFC 2833) and SIP INFO
Voice processing features
Voice activation detection, comfort noise generator, acoustic echo cancellation, packet loss concealment, adaptive jitter buffer, automatic gain control, full-duplex hands-free speakerphone with acoustic echo canceller.
Phone Features
- 6 VoIP accounts
- Call hold, mute, DND
- One-touch speed dial, hotline
- Call forward, call waiting, call transfer
- Group listening, SMS
- Redial, call return, auto answer
- Local 3-way conferencing
- Direct IP call without SIP proxy
- Ring tone selection/import/delete
- Set date and time manually or automatically
- Dial plan
- XML Browser
- Action URL/URI
- RTCP-XR (RFC3611), VQ-RTCPXR (RFC6035)
Directory
- Local phonebook up to 1000 entries
- Black list
- XML/LDAP remote phonebook
- Intelligent search method
- Phonebook search/import/export
- Call history: dialed/received/missed/forwarded
PBX Features
- Busy Lamp Field (BLF)
- Bridged Line Appearance (BLA)
- Anonymous call, anonymous call rejection
- Hot-desking, emergency call
- Message Waiting Indicator (MWI)
- Voice mail
- Call park, call pickup
- Intercom, paging
- Music on hold
- Call completion
- Call recording
Display and Indicator
- 320x160-pixel graphical LCD with backlight
- LED for call and message waiting indication
- Dual-color (red or green) illuminated LEDs for line status information
- Intuitive user interface with icons and soft keys
- National language selection
- Caller ID with name, number
Feature Keys
- 48 keys including 16 freely programmable keys
- 8 feature keys: message, headset, hold, mute, tran, redial, conf, hands-free speakerphone
- 4 context-sensitive "soft" keys
- 6 navigation keys
- Volume control keys
Interface
- 2 x RJ45 10/100M Ethernet ports
- Power over Ethernet (IEEE 802.3af), Class 2
- 1 x RJ9 (4P4C) handset port
- 1 x RJ9 (4P4C) headset port
- 1 x RJ12 (6P6C) EXT port: Supports up to 6 Expansion Modules for an attendant console application
Hardware Specifications
The following specifications are for the Japan market.
- Phone ports: handset port x 1, headset port x 1
- WAN/LAN: Two-port 10/100M Ethernet Switch
- LCD screen: 320 x 160-pixel graphical LCD with backlight
- Software function access keys: 48 keys including 16 freely programmable keys
- Wall mount bracket: not specified
- External universal AC adapter (optional): AC 100~240V input and DC 5V/1.2A output
- Power consumption (PSU): 1.3-2.1W
- Power consumption (PoE): 1.8-3.1W
- Dimensions (W x D x H x T): 273mm x 200mm x 158mm x 58.5mm
- Operating humidity: 10~95% RH
- Operating temperature: -10°C to 50°C
Management
- Configuration: browser/phone/auto-provision
- Auto provision via FTP/TFTP/HTTP/HTTPS for mass deploy
- Auto-provision with PnP
- Zero-touch, TR-069
- Phone lock for personal privacy protection
- Reset to factory, reboot
- Package tracing export, system log
Network and Security
- SIP v1 (RFC2543), v2 (RFC3261)
- Call server redundancy supported
- NAT traversal: STUN mode
- Proxy mode and peer-to-peer SIP link mode
- IP assignment: static/DHCP/PPPoE
- HTTP/HTTPS web server
- Time and date synchronization using SNTP
- UDP/TCP/DNS-SRV (RFC 3263)
- QoS: 802.1p/Q tagging (VLAN), Layer 3 ToS/DSCP
- SRTP for voice
- Transport Layer Security (TLS)
- HTTPS certificate manager
- AES encryption for configuration file
- Digest authentication using MD5/MD5-sess
- OpenVPN
- IEEE 802.1X
- IPv6
Package Features
- Qty/CTN: 5 PCS
- N.W/CTN: 6.9 KG
- G.W/CTN: 7.6 KG
- Gift Box size: 285mm x 240mm x 110mm
- Carton Measurements: 580mm x 295mm x 250mm