Grandstream DP715 VoIP DECT Phone - Base Station and Handset

The Grandstream DP715 VoIP DECT Phone is part of a new range of DECT VoIP Phones offered by Grandstream. The Phones have many features.

The Grandstream DP715 is the next generation of powerful, affordable, high quality and simple to configure VoIP DECT phone for small business and residential users.

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

• DECT base station 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

Key Features and Functions of the Grandstream DP715 VoIP DECT Phone

• 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 option may flexibly support:
• Linear Mode, all phones ring sequentially in the predestinated order
• Parallel Mode, all phones ring concurrently and after one phone answers the rest phones may place new calls
• Shared Line Mode, all phones ring concurrently and always share the same line similar to an analog phone
• 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
• Currently Pending - TR069, IPV6, Skype

Technical 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 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

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

• 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)