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深圳市錦辰科技有限公司


VIP-1010PT

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更新時間:2022-03-14 16:34:41瀏覽次數:1100次

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產品簡介

Cost-effective,High-performancePoEVoIPPhoneTobuildhigh-performanceVoIPcommunicationsatalowcost,PLANEThasintegratedhigh-definitionvoiceintoacost-effectiveSIPphone

詳細介紹

Cost-effective, High-performance PoE VoIP Phone
To build high-performance VoIP communications at a low cost, PLANET has integrated high-definition voice into a cost-effective SIP phone. It complies with IEEE 802.3af PoE interface for flexible deployment. The VIP-1010PT makes it simple for the enterprise featuring voice and data system or expanding voice system to new locations. It helps the company to save money on long-distance calls; for example, the remote workers can dial in through a Unified VoIP Communication System just like an extension call but no long-distance call charge would occur. The VIP-1010PT also allows call to be transferred to anyone at any location within the voice system, which enables the enterprise to communicate more effectively and is helpful to streamline business processes.
High-quality HD VoIP Voice
The VIP-1010PT delivers HD voice (High-definition Voice) which is the next generation of voice quality for telephony audio, making the quality of voice better than that (toll quality) of the standard digital telephony and even close to that of a room conversation. HD voice is transmitted in the audio frequency range of 50 Hz to 7 kHz or higher over telephone lines, resulting in higher quality voice and clearer communication.
Standard Compliance
The VIP-1010PT supports Session Initiation Protocol 2.0 (RFC 3261) for easy integration with general voice over IP system. The VIP-1010PT is able to broadly interoperate with equipment provided by VoIP infrastructure providers, thus enabling them to provide their customers with better multi-media exchange services.
Enhanced, Full-Featured Business IP Phone
The VIP-1010PT is a full-featured, enhanced business IP Phone that addresses the communication needs of the enterprises. It provides 1 voice line and dual 10/100Mbps Ethernet. Furthermore, the VIP-1010PT delivers user-friendly design containing a 132x64 graphic LCD with white backlight.

The VIP-1010PT supports all kinds of SIP-based phone features including LDAP, Call Waiting, Auto Answer, Music on Hold, Caller ID 3-way Conferencing, Call on Hold, Call Forwarding, Black List, DTMF Relay, In-Band, Out-of-Band (RFC 2833) and SIP Info, among others. Besides office use, the VIP-1010PT is also the ideal solution for VoIP service offered by Internet Telephony Service Provider (ITSP).
Enterprise IP Telephony Deployment of VIP-1010PT
The VIP-1010PT is much easier to install and configure than the traditional phone system. Its low cost and high-definition voice quality give you value for money. Base on standard SIP 2.0, it is compatible with all the standard SIP-based servers.
Highlights
    Supports SIP 2.0 (RFC3261) Supports 1 SIP voice lineIEEE 802.3af Power over Ethernet compliantSupports HD voiceLDAP/ TR-069 / SNMP

Phone Features
    1 line (supporting 1 SIP account)Supports call waiting, call forwarding, call transfer3-way conferencingCall on hold, mute, auto-answer, redialPhonebook (500 groups), blacklist (100 groups), call logs (100 entries)5 remote phone book URL supportedKeypad LockDND (Do Not Disturb)Volume adjustable, ring tones selectableCall Pickup/Group Call PickupSpeed DialIntercomDaylight SavingNetwork Packet CaptureCountry Ringtone SignalDirect IP CallAuto Redial / Hot Desking Hotline / XML Browser / Action URLMulti-Languages: Default: English and Simple Chinese

IP PPX Feature
    HD VoiceDial PlanSMS, Voic, MWI Message NotificationWideband Codec: G.722Narrowband Codec: PCMA, PCMU, G.729, G.722, G723_53, G23_63,G726_32VAD, CNG , Echo CancellerFull-Duplex Speakerphone

Security Features
    Supports HTTPS (SSL)Supports SRTP for Voice Data EncryptionSupports Login for AdministrationSIP Over TLS

Network Features
    SIP V1 (RFC2543), V2 (RFC3261)Static IP/DHCP for IP configuration3 DTMF modes: In-Band, RFC2833, SIP INFOHTTP/HTTPS Web Server for ManagemenP for Auto Time Setting

Administration Features
    Auto provisioning using FTP/TFTP/HTTP/HTTPS/PnPDial through IP PBX using Phone NumberDial through IP PBX using URL Address Configuration Managements with Web, Keypad on the phone and Auto ProvisioningSNMPTR069
硬件特性
1線1-line cost-effective IP phone
顯示屏132 x 64 graphic LCD with blue backlight
功能鍵4 Soft Keys
10 Programmable Keys
協議與標準
數據網絡技術MAC Address (IEEE 802.3)
IPv4 (RFC 791)
Address Resolution Protocol (ARP)
DNS: A record (RFC 1706), SRV record (RFC 2782)
Dynamic Host Configuration Protocol (DHCP) client (RFC 2131)
TCP (RFC 793)
User Datagram Protocol UDP (RFC 768)
Real-time Protocol RTP (RFC 1889, 1890)
Real-time Control Protocol (RTCP) (RFC 1889)
Simple Network Time Protocol (SNTP) (RFC 2030)
Backward compatible with RFC 2543
Session Timer (RFC 4028)
SDP (RFC 2327)
語音網關SIP version 2 (RFC 3261, 3262, 3263, 3264)
Message Waiting Indicator (RFC 3842)
Voice algorithms:
- PCMA
- PCMU
- G.729
- G.722
- G723_53
- G23_63
- G726_32
Dual-tone Multi-frequency (DTMF), In-Band and Out-of-Band (RFC 2833) (SIP INFO)
Voice Activity Detection (VAD) with Silence Suppression
Comfort Noise Generation
Echo Cancellation Message
功能特性
電話特性1 line (supporting 1 SIP account)
Supports call waiting, call forwarding, call transfer
3-way conferencing
Call on hold, mute, auto-answer, redial
Phonebook (500 groups), blacklist (100 groups), call logs (100 entries)
5 Remote Phone Book URL supported
LDAP
DND (Do Not Disturb)
Volume adjustable, ring tones selectable
Call Pickup/Group Call Pickup
Speed Dial
Intercom
Daylight Saving
Network Packet Capture
Country Ringtone Signal
Direct IP Call
Auto Redial
Hotline
XML Browser
Hot Desking
Keypad Lock
Action URL
Multi-Languages: Default: English and Simple Chinese
IP PBX 特性HD Voice
Dial Plan
SMS, Voic, MWI Message Notification
Wideband Codec: G.722
Narrowband Codec: PCMA, PCMU, G.729, G.722, G723_53, G23_63, G726_32
VAD, CNG, Echo Canceller
Full-Duplex Speakerphone
安全特性Supports HTTPS (SSL)
Supports SRTP for Voice Data Encryption
Supports Login for Administration
SIP Over TLS
Dial without Register
網絡特性SIP V1 (RFC2543), V2 (RFC3261)
Static IP/DHCP for IP configuration
3 DTMF modes: In-Band, RFC2833, SIP INFO
HTTP/HTTPS Web Server for Management
NTP for Auto Time Setting
管理功能Auto provisioning using FTP/TFTP/HTTP/HTTPS/PnP
Dial through IP PBX using Phone Number
Dial through IP PBX using URL Address
Configuration Managements with Web, Keypad on the phone and Auto Provisioning
SNMP
TR069
環境
Power Requirements5V DC, 1.2A
IEEE 802.3af
Operating Temperature0 ~ 40 degrees C
Operating Humidity10 ~ 65% (non-condensing)
Weight 651g (without box) / 920g (with box)
Dimensions (W x D x H)193 x 190 x 35 mm
EmissionCE, FCC
ConnectorsTwo 10/100BASE-TX RJ-45 Ethernet ports
Handset: RJ-9 connector
Headphone: RJ-9 connector
DC power jack
Built-in speakerphone and microphone


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