Description
Grandstream DP715 Dubai
The Grandstream DP715 Â 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) |