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snom 320

snom320

The Affordable Quality Business Phone

The snom 320 is an affordable, yet powerful high-end SIP business telephone with built-in, full-duplex speakerphone, three-party conference bridging and an ergonomic high-end handset and keyboard, ideal for general office, call center and knowledge-worker environments.

The snom 320 is remote-manageable and firmware-upgradeable, uniquely easy to install, and largely self-configuring. Broad codec support and full compatibility with current SIP recommendations insures interoperability; support for STUN (for NAT traversal), ENUM (for dialed-number resolution) and other state-of-the-art features enables flexible deployment, behind local proxies, IP PBXs or hosted VoIP services.

A 2 x 24 semi-graphic LCD display and menu-driven user interface supports custom branding and easy feature management. And the 320's built-in web server supports even simpler end-user configuration, screen dialing, and access to call history.

A 100-number call memory, 100-number onboard address book (to which data may easily be uploaded), custom call blocking, configurable/downloadable ring-tones, auto-answer mode, DND and other sophisticated features insure convenience and productivity.

12 programmable keys can be used to customize the functionality according to your own specific needs. The LED behind the corresponding function keys can show you whether or not your colleagues are currently on a call. And of course your colleagues can see whether your line is free or not.

To spare you the annoyance of unwanted invasions on your speech data, the snom 320 supports the security standard SRTP - a current specification from the Internet Engineering Task Force (IETF) for protection against eavesdropping and the stealing of data- and TLS for protection against sniffing of signaling and authentication data.

By limiting need for external conference bridges/media server capacity or use of conference services for routine multiparty calls, the snom 320's built-in three-party conference bridge helps limit total cost of ownership, while also insuring high audio quality and low latency.

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.

With its VoIP phones and software systems, snom ensures customers comfort, quality, open standards, and cost effectiveness.

Contact ABP to learn more about the snom phones and interoperability with softswitches and IPBXs.

 

Feature List
Technical Specifications
Product Applications Scenarios
Downloads

Feature List

2 x 24 character display

12 programmable function keys

2 port Ethernet switch

Speakerphone (Full Duplex)

PoE in WAN port

menu driven user interface

NAT-support , STUN client (NAT traversal)

Clock, daylight saving, call-timer

call hold, call wait

Call blocking (Deny List)/ Do not disturb

Auto answer mode

Conference call (3 way conference)

blind and attended transfer

call divert, call intrusion, call completion

Call park, call pick up

number guessing

speed dialing, handsfree-mode dialing

Address book (100 entries), Address book Import/Export

redial

missed calls, dialed calls (100 entries)

call waiting indication

Selectable ringing melodies/Downloadable ringing melodies

music on hold support

http / https server/client

Automatic software update

Completely automatic installation from web

national language support (NLS), Asian Language Support (ALS): Japanese

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Technical Specifications

Hardware

Power supply: 5V DC Stabilized

Memory: 16MB DRAM, 4MB FLASH

CPU: Infineon

Temp range: 0 C to 40 C (32°F to 104°F)

Measures: 25 cm W x 20 cm H x 12 D cm (10 x 8 1/4 x 4 3/4”), Weight: 860

Display: 2 x 24 display w/ graphical field

Keyboard: 27 keys, 7 LEDs, 6 programmable keys

Audio: CNG- Comfort Noise generation, VAD- Voice Activity detections Dynamic Jitter-Buffer, Loudspeaker, Inclusive 27 ring tones Voice compression, Echo compression, Silence suppression

Connector:
Network: (2) 10/100 BaseT E auto-detect RJ45 (Ethernet), Handset: RJ14 Standard
Headset: Standard Computer and RJ Connector

Software

Operating System: Linux

Configuration: Via web browser & menu

Protocols Internet: IP, TCP, UDP, DHCP, TLS, SIP, SIPS, SRTP

VoIP: Static IP provisioning, DHCP, SIP RFC3261 compliant, Digest authentication, Loose routing and strict routing support, Error-information support, VLAN (802.1 P/Q),
Reliability of provisional responses (RFC3262), DNS SRV (RFC3263), redundant server support, Offer/answer (RFC3264), Message waiting indication reception (RFC3842), MWI events (RFC3265), Dialog-State, STUN client (NAT traversal), ΕNUM (RFC3261), ΝAPTR (RFC2915), rport (RFC3581), REFER (RFC3515), Outbound proxy support, initial route

Voice encoding: G.711 aLaw, uLaw, G.722 (16Khz), G.723.1, G.726, G.729A, GSM 6.10 (Full rate), In-band DTMF, Out-of-band DTMF (RFC2833)

Security: HTTPS (server/client), Transport Layer Security (TLS), SRTP (RFC3711), Certificates can be loaded into device

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Product Application Scenarios

Enterprise Solutions

Branch Office Scenarios

Multi Tenant / Dwelling Scenarios

Government Scenarios

Educational Scenarios

Event Telephony Scenarios

Satellite Telephony Scenarios

IPBX OEM Partners

Downloads

Data Sheet 89KB

Quick Installation Guide 95KB

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