Grandstream DP715 Dubai

Grandstream DP715

The Grandstream DP715 Dubai 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 Dubai

The Grandstream DP715  Dubai VoIP DECT Phone system is a next generation high quality and simple to use DECT Cordless IP phone. It features compact size, superb voice quality, a rich feature set, and market leading performance while still being very affordable. This DECT Phone has wide range radio coverage, 150 feet indoors and up to 1000 feet outdoors, allows users to enjoy the benefits of mobility and VoIP for a minimum investment. It is also fully complaint with SIP/DECT standards and field proven for flexible deployments.

The DP715 handset has an indoor range of 150 feet and an outdoor range of about 1000 feet. outdoors, depending on blocking structures. With a standby time of 80 hours and up to 10 hours of talk time, the Grandstream DP715 system has what it takes to bring excellent communication to your corridor mobile workforce.


Grandstream DP715 Features and Functions

  • Includes DECT base station and one handset
  • Registers up to 4 additional DECT handsets
  • All phones can ring sequentially in the predestinated order
  • Shared Line Mode
  • Advanced telephony features
  • 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-Language Settings


Grandstream DP715 Dubai Specification

Air Interfaces Telephony standards: DECT / GAP Frequency range: 1880 – 1900 MHz (Europe), 1920 – 1930 MHz (US) Number of channels: 120 (Europe), 60 duplex (US) Modulation: GFSK Speech coding: 32 kbit/s Emission power: 10 mW (average power per channel) Range: up to 300 m outdoors, maximum of 50 m in buildings
Networking Interfaces One 10/100Mbps auto-sensing Ethernet port (RJ45) ( DP715 Base Station only)
LED Indicators Base Station : Power, Network, Register, Call
Handset Display 1.7″ 102×80 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
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. 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 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.
Multi-language Input English; Latin; Greek; Russian
Polyphonic Ringtones 18 different ringer melodies are available to indicate an incoming call (internal intercom or external VoIP)