Snom 370 IP Telephone in Black

Black Snom 370 SIP IP Telephone

High audio quality, ease of use, security and interoperability make the snom 370 desk phone 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.

Through the mini browser, users can have direct access to their own applications via the display screen. Users can not only customize the design of the display but, also view news tickers and other information as well as access central or public phone directories.  

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 SIP Telephone supports several audio devices simultaneously, for example, it is possible to use the handset, headset, and loudspeaker concurrently. 

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. 

Afraid of unwanted listeners, data theft or spam calls?

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. 

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.

Warranty

1 Years Manufacturers Warranty

Technical Specifications:

General Features

• Dimensions:approx. 25 x 20 x 13 cm

• Weight: approx. 960 g

•  Safety: IEC 60950-1:2001,CB Test

• Certifications: FCC Class B, CE Mark

• Power consumption: 2.6-2.8 watts

Connectors

•  1 x LAN, 1 x PC: RJ45 (Ethernet)

• Power: 5 V DC

• Ethernet: 2 x IEEE 802.3 10/100 Mbps switch

• ower over Ethernet:IEEE 802.3af, Class 1

• Handset: RJ11 connector

• Headset: RJ11 connector

• Expansion Module:Proprietary snom connector

User Interface

• 240 x 128 pixels, tiltable grayscalegraphical display, backlit

• 47 keys, 12 programmable function keys with LEDs (54 with the expansion module)

• Caller indication LED

• Caller-ID (with picture display)

• Message waiting indication LED

• Address book (250 entries)

• Address book import/export

• Speed dialing

• Local dial plan

• Number guessing

• Lists of missed, received and dialed calls (100 entries each)

• Call waiting indication

• Clock, daylight saving, call-timer

• Call blocking (deny list)

• Blocking of anonymous calls

•  Handling of up 12 calls simultaneously

• Menu-driven user interface

• Ring tone selection, import of individual ring tones

• National language support for selected languages (NLS)

• Do not disturb

• Speakerphone (full duplex)

•  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 pick-up (only via PBX)

• Call completion

• Client Matter Code (CMC)

• Call waiting/switching between calls

• Redialing

• RTP multicast paging

• Multiple audio device support

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, Quality of service

• HTTPS (server/client)

• Transport Layer Security (TLS)

• SRTP (RFC3711), SIPS

• RTCP

• VLAN (802.1 pq)

Codecs, Audio

• G.711 A -law, µ-law

• G.729A, G.723.1, G.727

• GSM 6.10 (full rate)

• G.722

• Comfort noise, voice activity detection

SIP

• RFC3261 compliance

• UDP, TCP and TLS

• Digest/basic authentication

• Loose routing and strict routing support

• PRACK (RFC3262)

• Error-information support

• Reliability of provisional responses(RFC3262)

• Early media support

• DNS SRV (RFC3263),redundant server support

• Offer/answer (RFC3264)

• Message Waiting Indication(RFC3842), subscription for MWI events (RFC3265)

• Dialog-state monitoring (RFC 4235)

• In-band DTM F/out-of- band DTM F/SIP INFO DTMF

• STUN client (NAT traversal)

• ENUM (RFC3261), NAPTR (RFC2915), rport (RFC3581),REFER (RFC3515)

• Event list subscription support (RFC4662)

• Bridged line appearance (BLA)

• Auto provisioning with PnP

• Presence/buddylist features

• 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