Our Price | $67.50 | |
MSRP | $85.00 | |
DP715 is the next generation of powerful, affordable, high quality and simple to configure VoIP DECT phones for small business and residential users.
Please Note: This model includes a base station.
Grandstream DP715 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. DP715 is SIP and DECT compliant and field proven for flexible deployment.
Grandstream DP715 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 DP715 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” 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 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) |