Grandstream DP715 VoIP DECT Mobile Phone and Base Station

The Grandstream DP715 is the next generation of
powerful, affordable, high quality and simple to configure VoIP DECT phones for
small business and residential users. Their compact size, superb voice quality,
rich feature set, market leading price-performance and wide range radio
coverage enable consumers to maximize the power of IP voice application and
mobility for a minimum investment. DP715/710 is SIP and DECT compliant and
field proven for flexible deployment.

DECT
base station (included with DP715) registers up to 5 DECT handsets and talks to
up to 4 handsets concurrently

Advanced
telephony features including Caller ID, Call Waiting, 3-Way Conference,
Transfer, Forward, Do Not Disturb, Message Waiting Indication(Stutter Tone),
auto answer, multi-language voice prompt, flexible dial plan

Secure
and automated provisioning using HTTP/HTTPS/Telnet/TFTP, multiple SIP accounts,
SIP over TCP/TLS, SRTP

Grandstream
DP715 Specifications

Air
Interfaces-Telephony standards: DECT / GAP Frequency range: 1880
- 1900 MHz (Europe), 1920 - 1930 MHz (US)
Number of channels: 120 (Europe), 60 duplex (US) channels
Emission power: 10 mW (average power per channel)
Range: up to 300m outdoors/50m indoors

Networking
Interface-One 10/100Mbps auto-sensing Ethernet port (RJ45) (
DP715 Base Station only)

LED
Indicators -Base Station : Power, Network, Register, Call

Handset
Display-1.7” 102x80 FSTN LCD with color backlight

Factory
Reset Button-Yes ( DP715 Base Station only)

Audio
Interface-Handsfree speaker (Handset only)

Voice
over Packet Capabilities-Base Station : Dynamic Jitter Buffer
Handset : Speakerphone with Acoustic Echo Cancellation

Voice
Compression-G.711 with Annex I (PLC) and Annex II (VAD/CNG),
G.723.1, G.726-32 AAL2, G.729A/B, iLBC

Telephony
Features-Caller ID display or block, call waiting, Flash,
blind or attended transfer, forward, hold, do not disturb, 3-way
conference/td>

QoS-Layer
2 (802.1Q VLAN/802.1p), Layer 3 (ToS, DiffServ, MPLS)

IP
Transport-RTP/RTCP

DTMF
Method-In-audio, RFC2833 and/or SIP Info

IP
Signaling-SIP (RFC 3261)

Multiple
SIP accounts per base station-Up to five (5) distinct SIP
accounts per system; Independent SIP account per handset; Multiple handsets per
SIP account

Hunting
Group-Linear mode; Parallel mode; Shared Line mode

Provisioning-HTTP,
HTTPS, TELNET, TFTP, TR-069 (pending), secure and automated provisioning

Security-Security
protection: SIP over TLS and SRTP.

Device
Management-Web interface or secure (AES encrypted) central
configuration file for mass deployment

Device
Management-Web interface or secure (AES encrypted) central
configuration file for mass deployment. Support device configuration via
built-in IVR, Web browser or central configuration file through TFTP, HTTP or
HTTPS. Auto/manual provisioning system. NAT-friendly remote software upgrade
for deployed devices including behind firewall/NAT. Syslog support

Phonebook(Per
Handset)-200 numbers (up to 24 digits) with an associated name
(up to 16 characters); 10 outgoing call entries; 30 incoming calls entries

Multi-language
Display-Base Station Web UI: English; Voice Prompt : English,
Spanish; Handset LCD Menu (15): English, French, German, Spanish, Dutch,
Italian, Czech, Danish, Greek, Norwegian, Polish, Portuguese, Russian, Swedish,
Turkish.

Polyphonic
Ringtones-18 different ringer melodies are available to
indicate an incoming call (internal intercom or external VoIP)

Battery
Life-10-hour talk time, 80-hour standby time, 16-hour charging
time

Box Content-DP715:
-1 Base Unit
-1 handset including 2 rechargeable batteries(AAA, 1,2V/500mAh, NiMH)
-1 power supply plug for base unit
-1 Ethernet cable
-service card
-user guide