Products VoIP ADSL Router LP600NSIP IP Phone for IP-PBX Application LP600NSIP IP Phone for IP-PBX Application Support 2 Simultaneous Calls at one IP-PBX Server account Rich Telephony Features Plug and Play to work with SIPPBX 6200x IP-PBX Server HTTP Provision provide auto configuration Intelligent Phone book Name dialing Support Multi-Party Voice Conference Color : Blue, Black and Grey Introduction LP600N is a SIP IP Phone which work as extension of SIPPBX 6200x series ( An IP-PBX server, for instance, SIPPBX6200A, SIPPBX6200S, SIPPBX6200GS, SIPPBX6200N and ePBX100A-128 ). LP600N IP Phone needs to configure those information includes Line number, Phone Book, IP-PBX or SIP Server, dial Plans and others refer to parameters to link with IP-PBX which can be configured by SIPPBX6200’s Administrator to fulfill without configuring LP600N one by one. The firmware upgraded to unique LP600N can be managed by SIPPBX6200x Administrator automatically. This convenient feature gives IP-PBX manager to configure office IP-PBX and LP600N features easily and effectively. Support 2 Simultaneous Calls LP600N Supports one Line number to register to one SIPPBX 6200x IP-PBX server and is able to make 2 phone calls. By using rich telephony feature, these IP-PBX features are available to increase your jobs efficiency. Plug & Play with SIPPBX 6200 IP-PBX Series LP600N related configurable features were stored at SIPPBX6200x IP-PBX server according to its MAC address before installation at customer site. This behavior gives office IP-PBX manager and user has enough time to verify his/her required feature and group planning to suit personnel demand before installation. Once LP600N installs at local site and link to DHCP server and SIPPBX6200x IP-PBX server automatically, it accepts to download its pre-configured personnel parameters and phone number as well. In a minute, LP600N is ready to use according to personnel configuration demand. LP600N’s installation is so easy and support Plug & Play without consuming a long installation time one by one. All those personnel information are stored at SIPPBX6200x IP-PBX server and can be exported to make a backup in case of hardware failure at SIPPBX 6200x or LP600N in order to resume your original configuration. Rich Telephony Features LP600N was designed to work with SIPPBX6200x IP-PBX and ePBX100A-128 (30 users small IP-PBX) in order to provide rich telephony PBX feature. For instance, Call Hold, Call Pick-up, Multi-party Conference, Call Transfer, Call Forwarding, Do Not Disturb, Phone book, Mute, Voice Mail, Headset, Missed calls, call records, use your preference Music as Ring and distinctive ring as well. Quick Phone Book name Dialing By using intelligent phone book name or phone number search engine inside, LP600N IP Phone user can dial number from their phone book easily by using the navigation keypad without using the difficult English character input from keypad. Support Up to 8 voice Parties Conference Calls LP600N itself supports voice mixer for 3-Way Voice Conference Calls. Besides, SIPPBX6200x IP-PBX server provides 8-Party Voice Conference Bridge. LP600N Switch Conference call Bridge to SIPPBX6200x Server automatically when it requires more than 3-Party conference calls. Specification Interface: Ethernet port (RJ-45, 10/100 base-T) 1-LAN port, for connecting to switch 1-PC port for connecting to PC 10/100 based Switch PoE (IEEE 802.3af ) at LAN port : LP600NA only Earphone/Microphone Jack (3.5mm) for Headset Handset Jack (RJ-10) DC 12V power input Jack LCD Display: Display Format: 16 Characters (W) x 2 lines (H) View Size: 64(W) x 17.9 (H) mm LCD Type: TN LCD Language Option: English, Chinese IP Network connection: IPv4 (RFC 791), MAC Address (IEEE 802.3) Static IP DHCP Client (RFC 2131) PPPoE Provide two DNS Server IP Address TCP/UDP (RFC 793/768) RTP/RTCP (RFC 1889/1890) IPV4 ICMP (RFC 792) TFTP Client VoIP VLAN Support (802.1Q/802.1P) VLAN ID Range : 2 to 4096 VLAN Priority : 0 to 7 (highest priority) HTTP/HTTPS Server QoS Support SIP Protocol : RFC3261 compliance Support 1 Line number, 2 calls at one SIP Register Account Support Primary and Backup SIP Proxy SIP Account Registration: Active, Auto Provision or Manual configure provision server. SIP Transport Type: UDP, TCP, TLS NAT Keep Alive Time SIP UDP Protocol Configurable SIP Local UDP, TCP and TLS port SIP QoS Type : DiffServ and TOS Voice RTP QoS Type : DiffServ and TOS Configurable Voice RTP port SIP Hold type Support SIP compact Form SIP Session Timer (RFC 4028) SIP Timer MD5 Digest Authentication Reliability of provisional responses PRACK (RFC3262) Early/Delay media support Offer/answer (RFC3264) Message Waiting Indication (RFC3842) Event Notification (RFC3265) REFER (RFC3515) Support DNS SRV to locate SIP Server (RFC 3263) Support STUN NAT Traversal Support “rport” parameter (RFC 3581) Audio Codec : G.711 A-law/μ-law, G.723.1 (6.3K/5.3K) G.729A, GSM 6.10 (full rate) Voice Codec Priority decision site : Local or Remote Voice Codec Payload Size ( ms ) configuration Silence Suppression VAD/CNG Adaptive/Configure Jitter Buffer AEC Tail Length (ms) configure Automatic Gain Control Preference Setting : Customized Idle Text display name at LCD Phone Book with desired incoming call Ring Tone or Music Intelligent Phone Book name Dialing Clock, Day Light Saving, Call-Timer Call History of Missed, Received and Dialed number Dial Plans Digit Manipulation ( DM ) Selectable Call Progress Tone Personal Music Ring Support Silence Ring Auto Answer Mode Call Features : 1 Line number under at one IP-PBX Server or SIP Server Caller ID display or inhibit Voice mail with Indication Speed Dialing Call Waiting/Switching between Calls Call Forward: Busy, Unconditional, No Answer Block Anonymous Call MIC and Volume configurable: Headset, Speaker, Handset, Ring In-band/out of band DTMF (RFC 4733 (RFC 2833)/SIP INFO) Configure RFC 2833 DTMF Payload Type Voice and SIP Command Encryption Send “REFER” without Hold Command 180 W/O SDP after RTP : Play Remote Voice or Play Ring Back Tone Program On-Net Call Telephone digits length Send DTMF before connect Program DTMF ON Time Do not disturb Call Hold Call Mute Call Transfer Call Forward : Busy, Unconditional, No Answer Block Incoming List phone number Music-on-hold support (via IPPBX6200x or local) 3-Way Conference (over phone) Server (IP-PBX 6200x) Conference Prefix code Multi-parties conference (via IPPBX 6200x) Distinct Ring between on-net and off-net calls (compatible with SIPPBX6200x) Call Pickup (via IPPBX 6200x) Call Park/Retrieve (via IPPBX 6200x) Voice Broadcasting (via IPPBX6200x) Barge-in & Barge-in Allowance List Voice an SIP Encryption Redialing/pre-dialing Hot Line : Dial pre-defined number immediately or manual dial within desired due time (second) Disable or Enable all features keys 3 User defined Keys to PSTN Line, Extension, Speed Dial or Speed dial with Input Text Inter Digit Time Out : 1 to 10 seconds Dial rule: Match dial prefix or Maximum digit Length Digit Manipulation (DM): Matched Prefix code Start digit Position Stop digit Position Replaced number MANAGEMENT : SNTP time server with Daylight Saving Variable Day, Month and Year display format HTTP/HTTPS and Telnet Command Enable or Disable HTTPS or Telnet Command Configurable port number of HTTPS and Telnet Multilingual Web User Interface Administrative Telnet CLI 3 Levels of User Access Right with Password protection and desired Web Language. Administrator Supervisor User Built-in Rich Debug feature Debug Phone Manager : Device Control, Call Control, Data Base Debug Phone level: Emergency, Alert, Critical, Error, Warning, Notice, Information, Debug. Debug SIP Manager: Register, SIP Message, Other Debug DSP SYSLOG Server to collect Debug messages Support HTTP Provision from SIPPBX 6200x and ePBX100A-128 IP-PBX Server LCD Administration Login Provides System Status Diagnostics (debug through syslog) Configuration Backup/Restore Reset to Default Dual Firmware Image Backup Auto provision by SIPPBX 6200x with Plug & Play Support HTTP provision through MAC address Central Management by SIPPBX 6200x IP-PBX Server: VLAN and DHCP Server provided by SIPPBX 6200x Extension Settings based on Mac address (up-to 3 Mac) Plug & Play without any settings on LP600N IP Phone Device-wide parameters Firmware Upgrade Time Settings Dial Plans Service Code Office/Personal Phone Book Environmental : Actual Dimension: 20 × 9.5 × 22.3 CM (Desktop) Weight: 1.1kg (with packing) Operating Temp. & Humidity Temp.: 0°C~45°C (32°F~113°F) Humidity: 10%~90% relative humidity, non-condensing Power Adaptor: INPUT: AC100V~240V, 50/60Hz OUTPUT: DC 12V, 1.0A Power Consumption of PoE : 4 Watts Approvals: CE, FCC, LVD and RoHS Country of origin: Made in Taiwan Packing Accessories LP600N IP Phone x 1 pcs AC to DC12V Power adaptor x 1 pcs CD User Manual Warranty One year Ordering Information: LP600N LP600NA PoE NO YES LCD Language English English IPv4 ONLY YES YES Headset, Ear Phone Microphone, Hand-Free YES YES Wall Mount NO NO Delivery Status NOW NOW Please write down your comment on this product, and click Add to Inquiry List for inquiry Add to Inquiry List Review Inquiry List Back