Product Overview

High audio quality, ease of use, security and interoperability make the snom 370 highly suitable for SMEs, home offices, private users or ISP applications. The snom 370 is the VoIP phone for the business user who needs immediate access to all of their important information. With the large graphical, high-definition display, the snom 370 offers an improved and extended presentation of call lists, address books and caller information. Caller information can be customized easily through XML to depict the information the user wants displayed.

Flexibility

More flexibility and productivity for your everyday work life. The improved interface guarantees ease of use and comfort. Context-sensitive menus offer additional options according to individual requirements. The snom 370 supports several audio devices simultaneously, for example, it is possible to use the handset and the loudspeaker or headset and loudspeaker concurrently.

Expanded Memory

More performance through expanded memory. The snom 370 offers more additional customer-oriented functionalities and applications. With its expanded memory capacity in addition to all necessary office functionality such as choice of trunk line, status indicator, group lines, transfer, call-pickup or conferencing (3-way conference bridge) more scope for individual functions and applications. The expanded memory capacity also enables the depiction of graphics and high resolution pictures to show the status of contacts (e.g., busy, on-line, off-line) similar to Instant Messenger. It is also possible to play music and media files and to integrate applications like VPN or snom CTI.

Security

To avoid problems with unwanted violations of your audio data, the snom 370 supports the security standards TLS (Transport Layer Security), SRTP, and SIPS which are necessary to protect against electronic eavesdropping and data theft. For IP-PBX that do not support secure communication, the snom 370 can employ VPN to tunnel the voice between the phone and the PBX. This also solves the problem with NAT for home users. With SIP (Session Initiation Protocol) you are ensuring your own personal independence. Most vendors are tipping SIP to be the communication protocol of the future. SIP components can be combined into a complete system without the need to be tied to a single provider.

Physical Specifications

Dimensions	250 x 200 x 135 mm
Weight	1kg
Certification	IEC 60950-1:2001 CB Test Certificate: DE 2-008417 FCC Class B CE Mark
Power Consumption	2.8 - 2.9 Watts

Phone Interface

External Interface

1 x LAN 1 x PC (RJ45) 1 x Power 2 x 10/100 Ethernet 1 x Handset RJ 4P4C Connector 1 x Headset RJ 4P4C Connector 1 x Expansion Module Connector

Phone Features

User Interface

- 240 x 128 pixels, tiltable grayscale display - 47 keys, 12 programmable function keys with LEDs - Call indication LED - Caller-ID (with picture display) - Message waiting indication LED - Address book (250 entries) - Speed dialing - Local dial plan - Number guessing - Lists of missed, received and dialed calls - Call waiting indication - Clock, daylight saving, call-timer - Call blocking (deny list) - Blocking of anonymous calls - Handling of up to 12 calls simultaneously - Menu-driven user interface - URL Dialing support - Ring tone selection, import of individual ring tones - National language support for selected languages - Do not disturb - Speakerphone - Auto answer mode - Keyboard lock

Call Features

- Hold - Blind transfer, attended transfer - Music-on-hold support (only via PBX) - Divert - Conferencing (3-way conference bridge on phone) - Call park, call pickup (only via PBX) - Call completion - Client Matter Code (CMC) - Call waiting/switching between calls - Redialing - RTP multicast paging - Multiple audio device support

Network Specifications

Web Server	- Embedded web server HTTP/HTTPS - Easy configuration of the phone, remote configuration - Dial from web interface - Password protection - Diagnostics (tracing, logging, syslog)
Security, QoS	- HTTPS (server/client) - Transport Layer Security (TLS) - SRTP (RFC3711), SIPS - RTCP, S-RTCP - VLAN (IEEE 802.1X) - LLDP-MED
Audio Codec	- G.711 A-law, µ-law - G.729A, G.723.1, G.726, G.722 - GSM 6.10 (full rate) - Comfort noise, voice activity detection
SIP	- RFC3261 compliance - UDP, TCP and TLS - Digest/basic authentication - PRACK (RFC3262) - Error-information support - Reliability of provisional responses(RFC3262) - Early media support - DNS SRV (RFC3263), redundant server support - Offer/answer (RFC3264) - MWI (RFC3842), MWI events (RFC3265) - Dialog-state monitoring (RFC 4235) - In-band DTMF/out-of-band DTMF/SIP INFO DTMF - STUN client, ICE (NAT traversal) - ENUM (RFC3261), NAPTR (RFC2915) - rport (RFC3581), REFER (RFC3515) - Bridged line appearance (BLA) - Auto provisioning with PnP - Presence/buddylist feature - Busy lamp field support (BLF) - Presence publishing
Installation	- Automatic software update - Automatic settings retrieval via HTTP/HTTPS/TFTP - Installation via web interface - Static IP provisioning, DHCP - NTP