Snom 320 2-Line SIP based IP phone
Ideal for the office and everyone who makes a lot of calls, the snom 320 is an affordable, yet powerful SIP business telephone with built-in full duplex speakerphone and three-party conference bridging.
How does it work?
Hailing from Germany, Snom is fast gaining ground in the SIP marketplace. Snom phones offer excellent interoperability with Asterisk Open Source PBX, Trixbox, PBXnSIP and other SIP based platforms.
The Snom 320 is Snom’s mid-level, SIP based IP phone. For under $200, you get an LCD display and programmable function keys, dual onboard ethernet ports, PoE multiple line appearances, full duplex speakerphone and a range of codec support.
A 2 x 24 semi-graphical LCD display and a menu-driven user interface supports easy feature management.
12 programmable keys with LEDs support flexible identity access/key configuration. A 100 number call
memory, a 100 entry onboard address book (to which data may easily be uploaded),custom call blocking, configurable/ downloadable ringtones,auto-answer mode, DND and other sophisticated features ensure convenience and productivity. And the snom 320’s built-in web server supports even simpler end user configuration,screen dialing, and access to call history.
The snom 320 is remote-manageable and firmware- upgradable, uniquely easy to install, and largely self-configuring.
Broad codec support and full compatibility with current SIP recommendations ensure interoperability;support for STUN (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.
The snom 320 supports the security standard SRTP – a current specification from the Internet Engineering Task Force (IETF) for protection against eavesdropping – and TLS for protection against sniffing of signaling and authentication data.
By limiting the 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 ensuring high
Highlights
- Tiltable semi- graphical two-line display
- 47 keys, 13 LEDs
- 12 programmable function keys
- Speakerphone
- 2 x IEEE 802.3 10/100 Mbps switch
- Power over Ethernet
- Headset connection
- SIP RFC3261
- Security: SIPS/SRTP, TLS
- STUN, ENUM, NAT, ICE,IEEE 802.1X
- Codecs: G.711, G.729A, G.723.1,G.722, G.726, GSM 6.10 (full rate)
- National language support
- XML driven mini browser
- Very low energy consumption
- Expansion module available
- Unified Communications ready
General Features
- Dimensions: approx. 25 x 20 x 12 cm
- Weight: approx. 920 g
- Certifications: FCC Class B, CE Mark
- Power consumption: 2.1 – 2.4 watts
Connectors
- 1 x LAN, 1 x PC: RJ45 (Ethernet)
- Power: 5 V DC
- Ethernet: 2 x IEEE 802.3 10/100 Mbps switch
- Power over Ethernet: IEEE 802.3af, Class 1
- Handset: RJ-4P4C connector
- Headset: RJ-4P4C connector
- Expansion Module: Proprietary snom connector
User Interface
- 2 x 24 character, tiltable semi-graphical display with backlit
- 47 keys, 12 programmable function keys with LEDs (54 with the expansion module)
- Caller-ID
- Message waiting indication LED
- Address book (100 entries)
- 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 to 12 calls simultaneously
- Menu-driven user interface
- Ring tone selection, import of individual ringtones
- URL Dialing support
- 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 pickup (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, S-RTCP
- VLAN (IEEE 802.1X)
- LLDP-MED
Codecs, Audio
- G.711 A-law, μ-law
- G.722, G.729A, G.723.1, G.726
- 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)
- Message Waiting Indication (RFC3842),subscription for 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
- Busy lamp field support (BLF)
Installation
- Automatic software update
- Automatic settings retrieval via HTTP/HTTPS/TFTP with authentication
- Installation via web interface
- Static IP provisioning, DHCP
- NTP
Snom 320 Solution Overview
The snom 320 features 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. And the 320\’s built-in web server supports even simpler end-user configuration, screen dialing, and access to call history.
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.
The snom 320 supports the security standard SRTP – a current specification from the Internet Engineering Task Force (IETF) for protection against eavesdropping – 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.