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Snom 320 IP Phone Handset.
Advanced, Affordable, Feature Rich
corporate style IP Phones for small to
medium size business.
Ideal for general
office and knowledge-worker environments, the snom 320 is an
affordable, yet powerful SIP business telephone with built-in,
full-duplex speakerphone and three-party conference bridging.
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12 programmable function keys
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47 keys, 13 LEDs
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Speakerphone
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Dual Ethernet
connection
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Power over
Ethernet
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Headset connection
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SIP RFC3261
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Security: SIPS/SRTP,
TLS
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STUN, ENUM, NAT,
ICE
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Codecs: G.711,
G.729A, G.723.1, G.722, G.726, GSM 6.10 (Full rate)
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National Language
Support
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Snom 320 $ 318.00 Inc
GST |
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Product Information
A 2 x 24
semi-graphic LCD display and menu-driven user interface supports
custom branding and easy feature management. 12 programmable keys
with LEDs support flexible trunk-access/busy lamp configuration.
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
dialling, 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
signalling 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
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Dimensions:
approx. 25 x 20 x 12 cm
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Weight:
approx. 920 g
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Certifications:
FCC Class B, CE Mark
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Power consumption:
typ. 2.3 Watt
Connectors
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Network:
RJ45 (Ethernet)
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PC:
RJ45 (Ethernet)
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Power:
5 V DC
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Ethernet:
2 x IEEE 802.3 10/100 Mbps
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Power over Network:
IEEE 802.3af (PoE)
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Handset:
RJ11 connector
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Headset:
RJ11 connector
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Expansion Module:
Proprietary snom connector
User Interface
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2
x 24 character semi-graphical display
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47
keys, 13 LEDs
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12
identities
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Call lists (100 entries)
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Address book (100 entries)
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Address book import/export
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Number guessing, speed dialing
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Missed calls, dialed calls, answered calls
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Call waiting indication
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Clock, daylight saving, call-timer
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Call blocking (deny list)
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Handling of 12 simultaneous calls
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12
programmable function keys. 54 with expansion module
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Menu-driven user interface
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Selectable ring tones
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Import of individual ring tones
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National language support for selected languages (NLS)
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URL dialling support
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Do
not disturb
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Speakerphone (full duplex)
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Auto answer mode
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Keyboard lock
Call
Features
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Hold
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Blind transfer, attended transfer
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Music on hold support (only via PBX)
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Divert
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Conferencing (3-way conference bridge on phone)
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Call park, call pick-up (only via PBX)
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Call completion
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Client Matter Code (CMC)
Web Server
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Embedded web server HTTP/HTTPS
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Easy configuration of the phone, remote configuration
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Dial from web interface
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Password protection
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Diagnostics (tracing, logging, syslog)
Security, Quality
of Service
CODEC
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G.711 A-law, ì-law
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G.722, G.723.1, G.726, G.729A, GSM 6.10 (full rate)
SIP
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RFC3261 compliant
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UDP, TCP and TLS
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Digest authentication
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Loose routing and strict routing support
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Error information support
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Reliability of provisional responses (RFC3262)
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DNS SRV (RFC3263), redundant server support
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Offer/answer (RFC3264)
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Message Waiting Indication (RFC3842, RFC3265)
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Dialog-state
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In-band DTMF/Out-of-band DTMF/ SIP
INFO
DTMF
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STUN client (NAT traversal)
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ENUM (RFC3261)
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NAPTR (RFC2915)
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rport (RFC3581)
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REFER (RFC3515)
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Many other SIP features
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Installation
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Automatic software update
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Automatic settings retrieval via HTTP/HTTPS/TFTP
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Installation via web interface
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Static IP provisioning, DHCP
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NTP
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Installation Options.
1) Install it yourself.
2)
Pre-Configured to your specifications.
3) Let the experts to install
it for you.
BlueWolf’s
Alfa team are fully trained to install VoIP PBX systems. We cater
for large commercial sites and small business offices.
Contact us
for free advice and an obligation free quote. |