Caller ID phone, VoIP telephone
The Q710 is an advanced, fully featured two-lines IP Telephone that takes full advantage of VoIP technology by offering a flexible, interoperable solution at an affordable price it leverages the SIP standard to provide advanced calling, conferencing, and messaging features, and protects investments with an upgradeable flash-based architecture
Feature Highlights:
1. Two lines / call appearances with LEDs
2. Conference calling (with or without server)
3. Call transfer (blind or consultative)
4. Call hold, call waiting, and call forwarding
5. Caller ID and Call-waiting caller ID
6. Do not disturb
7. Number note pad
8. Paging
9. Auto answer
10. Configurable dial plan
11.500 entry phone book
User interface:
Keypad dialing
Speed dial from phone book and call log
100 number redial list
Menu layout for easy navigation via softkey
Missed / outgoing / incoming call logs, 100entries each voicemail retrieval via keypad or programmable keys
8 distinctive ring tone
Volume and Ringer level control
4 or 12 (with expansion module) programmable keys for speed dial or feature access.
Configuration and Management:
Simple Web-based configuration and status check
Auto-provisioning using TFTP
Update phone book remotely via web browser
Software upgrade via TFTP or FTP
Enterprise Functionality:
Integrated 2 port 10/100 Mbps Ethernet switch
Hands free full-duplex speaker phone
802.3af Power over Ethernet
Headset jack (RJ9)
VLAN and DSCP support
Technical Specification
Protocols Supported:
SIP (RFC3261, RF3262)
SDP (RFC2327)
(SIP) Subscribe / Notify (RFC3265)
(SIP) Offer / Answer Model (RFC3264)
(SIP) Refer Method (RFC3515)
DTMF transport (RFC2833)
SIP INFO and in-band DTMF transport
RTP / RTCP (RFC3550, 3551)
(SIP) DNS SRV (RFC2781)
IAX2
Peer to Peer calling
Networking:
WAN / LAN: Support Bridge and Router mode
Support basic NAT and NAPT
Support PPPoE for xDSL.
Support DHCP Client on WAN
Support DHCP Server on LAN
Support DNS Relay, SNTP Client, Firewall
Network tools in telnet server: Including ping, race route, telnet client
Qos:
DiffServ / IP Precedence; IEEE802.3p
Audio / Codec:
G. 711 A law and G. 711 U law 64K
G. 729
G. 723.1
DTMF tone Generation
Acoustic echo cancellation
Built-in audio mixer
Security:
Two level password access
User level configuration and Status:
Brower based configuration interface and status page
Edit phone book and programmable keys using web interface
LCD display for access to configuration and customized setting
The Q710 is an advanced, fully featured two-lines IP Telephone that takes full advantage of VoIP technology by offering a flexible, interoperable solution at an affordable price it leverages the SIP standard to provide advanced calling, conferencing, and messaging features, and protects investments with an upgradeable flash-based architecture
Feature Highlights:
1. Two lines / call appearances with LEDs
2. Conference calling (with or without server)
3. Call transfer (blind or consultative)
4. Call hold, call waiting, and call forwarding
5. Caller ID and Call-waiting caller ID
6. Do not disturb
7. Number note pad
8. Paging
9. Auto answer
10. Configurable dial plan
11.500 entry phone book
User interface:
Keypad dialing
Speed dial from phone book and call log
100 number redial list
Menu layout for easy navigation via softkey
Missed / outgoing / incoming call logs, 100entries each voicemail retrieval via keypad or programmable keys
8 distinctive ring tone
Volume and Ringer level control
4 or 12 (with expansion module) programmable keys for speed dial or feature access.
Configuration and Management:
Simple Web-based configuration and status check
Auto-provisioning using TFTP
Update phone book remotely via web browser
Software upgrade via TFTP or FTP
Enterprise Functionality:
Integrated 2 port 10/100 Mbps Ethernet switch
Hands free full-duplex speaker phone
802.3af Power over Ethernet
Headset jack (RJ9)
VLAN and DSCP support
Technical Specification
Protocols Supported:
SIP (RFC3261, RF3262)
SDP (RFC2327)
(SIP) Subscribe / Notify (RFC3265)
(SIP) Offer / Answer Model (RFC3264)
(SIP) Refer Method (RFC3515)
DTMF transport (RFC2833)
SIP INFO and in-band DTMF transport
RTP / RTCP (RFC3550, 3551)
(SIP) DNS SRV (RFC2781)
IAX2
Peer to Peer calling
Networking:
WAN / LAN: Support Bridge and Router mode
Support basic NAT and NAPT
Support PPPoE for xDSL.
Support DHCP Client on WAN
Support DHCP Server on LAN
Support DNS Relay, SNTP Client, Firewall
Network tools in telnet server: Including ping, race route, telnet client
Qos:
DiffServ / IP Precedence; IEEE802.3p
Audio / Codec:
G. 711 A law and G. 711 U law 64K
G. 729
G. 723.1
DTMF tone Generation
Acoustic echo cancellation
Built-in audio mixer
Security:
Two level password access
User level configuration and Status:
Brower based configuration interface and status page
Edit phone book and programmable keys using web interface
LCD display for access to configuration and customized setting