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Grandstream DP710

Grandstream DP710 Summary

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. DP710 is SIP and DECT compliant and field proven for flexible deployment.

Grandstream DP710 Core Features

  • DECT base station registers up to 5 DECT handsets and talks to up to 4 handsets concurrently
  • When multiple handsets share the same SIP account, Hunting Group supports the following flexible options:
    • Linear Mode, all phones ring sequentially in the predestinated order 
    • Parallel Mode, all phones ring concurrently and after one phone answers,the remaining available phones can place new calls 
    • Shared Line Mode, all phones ring concurrently and always share the same line similar to analog phones
  • Advanced telephony features including Caller ID, Call Waiting, 3-Way Conference, Transfer, Forward, Do Not Disturb, Message Waiting Indication, auto answer, multi-language voice prompt, flexible dial plan
  • Support comprehensive voice codecs including G.711, G.723.1, G.729A/B, G.726 and iLBC
  • Secure and automated provisioning using HTTP/HTTPS/Telnet/TFTP, multiple SIP accounts, SIP over TCP/TLS, SRTP
  • Multi-Languages - English, German, French, Spanish, Dutch, Italian, Czech, Danish, Greek, Norwegian, Polish, Portuguese, Russian, Swedish, Turkish

 

Grandstream DP710 Technical 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)

 
End user Pricing
Unlimited stock
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