Stonehenge

Transcript

STONEHENGE IP250 /IP255 Field Proven VoIP Phone Widely used voip telephone in Market Stable and Deferentiable solution for provider or dealer IP250 / 255 SIP As VoIP phone for business based on standard SIP, it meets every function of various IP PBX and soft-switches embodying linked function enough. It proved their stability and VoIP technique by installing them into domestic and overseas sites. Based on the proved technique, it increase the convenience and productivity. 제품 특징 SIPv2(RFC3261) of VoIP Phone Excellent sound quality by various Codec As Global standard SIP internet phone, It allows easy installing and accessing Adapting various Codec including G.711a Law/µLaw, to anywhere connected with internet, SIP service server and installed soft G.279.A/B, guarantees high quality sound as much as switch. conference phone. IPv4/IPv6 dual stack Graphic LCD for user Supporting IPv4 and IPv6 together Graphic LCD in English helps user to be accessible to various function easily. RTP and Transport Layer Security Effective remote management For effective management, it supports Auto To avoid call hacking, it supports a high level signal Provision function which can easily upgrade assuring the excellent security. software and install phone. IP Phone FXO & PoE Optional PoE(Power of Ethernet): Self power adapter over Ethernet. IP Phone PoE Optional PoE(Power of Ethernet): Self power adapter over Ethernet. FXO(Foreign Exchange Office): Additional PSTN port www.moimstone.com STONEHENGE IP250 /IP255 Specification Call Function Voice & Codec Key & Button • Call Forwarding • Call Hold • Call screen/ Do Not Disturb • Call Pickup • Call Transfer • Call Waiting • 3 Party Conference call • SIMPLE based Instant Messaging and Presence • Auto Provisioning (HTTPS, HTTP, TFTP) • CNG (Comfort Noise Generation) • Echo Cancellation: G.168 compliance • Codec Auto Negotiation • Codecs - Narrow band -G.711μLaw/aLaw with PLC, G.722.1, G.723.1/A, G.726, G.728, G.729A/AB/E , Broadvoice®16 - Wide band – G.722.1, Broadvoice®32 • Echo Suppression (G.164) • Enhanced Packet Loss Concealment • Silence Suppression (G.164) • VAD (Voice Active Detector) • Adaptive Jitter Buffer • SIPFrag (RFC 3420) • Dynamic Payload Support • Adjustable Audio Frames per Packet • Flexible Dial Plan Support with Inter-Digit Timers • Dial Keys: 12 keys (ITU E.161) • 9 Function Keys: Redial, Transfer, 3 Way call, Call Pick up, Hold, Mute, Headset, DND, Call Forwarding • Speaker Phone Button for Handfree conferencing • Additional 3 Service Buttons: Do Not Disturb function, Call Forwarding, Message • 6 Hotkeys • 2 Volume Control Keys; can be used as navigation (up & down) Keys on the Menu • Phonebook: Record 100 Entries • Call history Logs: Record 60 Entries for In/ Outbound or Missed Calls SIP Protocal • Proxy Registration and Failover • Outbound Proxy • Multi-user Registration • Registration Timer • SIP Transport – UDP, TCP, TLS • Secured media negotiation(SRTP) • Realm-based authentication (Digest authentication) • Session Timers • DNS query (A record, SRV, NAPTR) • Codec Negotiation • DTMF relay RTP payload(RFC 2833) or SIP info • Hook flash signaling • Visual Message Waiting Indicator DTMF/Ring Signal • DTMF (Dual-Tone Multi-Frequency) • Multiple Ring Tones • My bell: 10 Ring • Call Progress Tone Generation (Dial tone, Busy, Audible Ring back) • DTMF generation (RFC 2833 In-Audio or Out-of-band SIP info) IP Network • IPv4 (RFC 791), IPv6(RFC1883) dual stack (optional), TCP, UDP, HTTP, ARP, ICMP • RTP/RTCP, Secure RTP • DNS: A record (RFC 1706), SRV record (RFC 2783) • NAT/PAT • VLAN: IEEE 802.1q • QoS: IEEE 802.1p, DiffServ(RFC 2475) • Network Address Assignment: Static IP/DHCP, PPPoE Display • Black and White LCD: 2x16 Character LCD or 128x64 Graphic LCD with Back Light Multi-Language Support: English + one • Ringing and Message Indicator: Green LED for status indication Power • Power Adaptor: Output DC5V/1A, rated Input AC100~240V. 50/60Hz • Power Consumption: 5Watts(max) • PoE (Power over Ethernet) *Optional Managemen Technical Spec • Password Protection for Admin mode and User mode • Management Protocol: SNMPv2 (RFC 2782) • Auto Provisioning: DHCP TFTP, Static TFTP, HTTP • Remote Software Upgrade: http,tftp • Remote Configuration • Measurements: 82x109x125mm (WxHxL) • Weight: 800g • Color: beige, black • Operating Temp: 0~45°C • Storage Temp: -20~60°C • Humidity: 10~85% (Non-Condensing) • Storage Temp: -20~60°C • Phone Stand • Handset: RJ-7 Standard Connector Ethernet • Dual switched 10/100 Based-T Through RJ-45 Interfaces • 10/100BASE-T: 2 Ports Providing auto-MDI/MDIX, Enabling the Use of Straight or Crossover Cable in Either Port • 1 RJ-45 Port to Connect LAN or Wan, this Port Supports Optional PoE (IEEE802.3af or In line power) • 1 RJ-45 port to Connect PC RFCs supported Optional • IP250: PoE or FXO • IP255: PoE(Power over Ethernet) IETF drafts supported RFC2327 Session Description Protocol(SDP) RFC3428 SIP Extension for Instance Messaging RFC2976 The SIP INFO Method RFC3515 The SIP Refer Method RFC3261 SIP: Session Initiation Protocol RFC3725 RFC3262 Reliability of Provisional Responses in SIP Best Current Practices for Third Party Call Control in SIP RFC3263 SIP : Locating SIP Servers RFC3842 A Message Summary and Message Waiting Indication Event Package for SIP RFC3264 An Offer/Answer Model with SDP RFC3892 The SIP Referred-By Mechanism RFC3265 SIP – Specific Event Notification RFC3903 SIP Extension for Event State Publication RFC3420 Internet Media Type message/sipfrag RFC4568 SDP Security Descriptions for Media Streams WWW.MOIMSTONE.COM Draft-ietf-sipping-cctransfer-01 SIP call control – Transfer Draft-ietf-sip-replaces-02 The SIP Replaces Header Draft-ietf-sip-sessiontimer-08 The SIP Session Timer Contact E-mail: [email protected] STONEHENGE IP270 SIP based VoIP Superset High Class IP phone based on standard SIP Office VoIP solution for many functions and simple user interface Stonehenge IP270 This IP270 is one of high class VoIP telephone based on Standard SIP, for convenient use, it support various function key as well as supplementary function. Graphic LCD and Bicolor LED help to facilitate the function. The proved the VoIP technique and stability of Moimstone from the market are the good solution. 20 Memory dials and menu button Graphic Icon on Menu 20 speed dials are available for user, and 4 direction keys and frequently used function key are available also. IP 270 shows environment for use as Graphic Icon. Those icons facilitate the PoE(Power over Ethernet) IP270 earns power from Network Ethernet line. So it doesn’t need additional power adapter. understanding for VoIP function. FXO for PSTN(Option) By building additional FXO(Foreign Exchange office) port optionally, it offers the change to PSTN from VoIP. PSTN backup Battery for power outage Backup Battery is built in the phone in case the abrupt power outage while using PSTN Line. With Backup Battery, LCD, LED and all functions will be operational Bicolor LED for BLF when power is out. BLF(Busy Lamp Filed) enables user to check if other lines previously installed are busy. Wired IP Communication l Wireless IP Communication l Conference Communication l IP Keyphone System www.moimstone.com STONEHENGE IP270 Specification Call Function Voice & Codec FXO • Call Forwarding • Call Hold • Call screen/ Do Not Disturb • Call Pickup • Call Transfer • Call Waiting • Intercom • 3 Party Conference call • SIMPLE based Instant Messaging and Presence • Auto Provisioning(HTTPS, HTTP, TFTP) • CNG (Comfort Noise Generation) • Echo Cancellation: G.168 compliance • Codec Auto Negotiation • Codecs - Narrow band -G.711μLaw/aLaw with PLC, G.722.1, G.723.1/A, G.726, G.728, G.729A/AB/E , Broadvoice®16 - Wide band – G.722.1, Broadvoice®32 • Echo Suppression (G.164) • Enhanced Packet Loss Concealment • Silence Suppression (G.164) • VAD (Voice Active Detector) • Adaptive Jitter Buffer • SIPFrag (RFC 3420) • Dynamic Payload Support • Adjustable Audio Frames per Packet • Flexible Dial Plan Support with Inter-Digit Timers • 1 RJ-11 FXO port for PSTN Use SIP Protocal • Proxy Registration and Failover • Outbound Proxy • Multi-user Registration • Registration Timer • SIP Transport – UDP, TCP, TLS • Secured media negotiation(SRTP) • Realm-based authentication(Digest authentication) • Session Timers • DNS query (A record, SRV, NAPTR) • Codec Negotiation • DTMF relay RTP payload(RFC 2833) or SIP info • Hook flash signaling • Visual Message Waiting Indicator DTMF/Ring Signal • DTMF (Dual-Tone Multi-Frequency) • Multiple Ring Tones • My bell: 10 Ring • Call Progress Tone Generation (Dial tone, Busy, Audible Ring back) • DTMF generation (RFC 2833 In-Audio or Out-of-band SIP info) IP Network • IPv4 (RFC 791), IPv6(RFC1883) dual stack (optional), TCP, UDP, HTTP, ARP, ICMP • RTP/RTCP, Secure RTP • DNS: A record (RFC 1706), SRV record (RFC 2783) • NAT/PAT • VLAN: IEEE 802.1q • QoS: IEEE 802.1p, DiffServ(RFC 2475) • Network Address Assignment: Static IP/DHCP, PPPoE Managemen • Password Protection for Admin mode and User mode • Management Protocol: SNMPv2 (RFC 2782) • Auto Provisioning: DHCP TFTP, Static TFTP, HTTP • Remote software upgrade: http,tftp • Remote Configuration Ethernet • Dual switched 10/100 Based-T through RJ-45 interfaces • 10/100BASE-T: 2 ports providing auto-MDI/MDIX, enabling the use of straight or crossover cable in either port • 1 RJ-45 port to connect LAN or Wan, this port supports Defaulted PoE (IEEE802.3af or In line power) • 1 RJ-45 port to connect PC RFCs supported Key & Button • Dial Keys: 12 keys (ITU E.161) • 11 Function Keys: Transfer, 3 Way call, Pick up, FXO, Hold, Redial(MUTE), Intercom, Menu, Headset, DND(ENT), Call Forwarding, Esc, Message(DEL) • Speaker phone button for hands free conferencing • Additional 20 Keys for Hot key • 4 navigation keys: Up&Down keys can be used for volume control • Phonebook: Record 100 entries • Call history Logs: Record 60 entries for In/ Outbound or Missed Calls Display • Black and White (Gray) LCD: 255x64 Graphic LCD with back light • Multi-Language Support: English + one • 20 LED: 4 special key + 16 user setting key - OO Bicolor LED for line status indication Power • PoE (Power over Ethernet) - Optional • Power Consumption: 5Watts(max) • Power Adaptor: Output DC5V/1A, rated input AC100~240V. 50/60Hz Technical Spec • Measurements: 252x205x112.5mm (WxHxL) • Weight: 1.2Kg • Color: beige, black • Operating Temp: 0~45°C • Storage Temp: -20~60°C • Humidity: 10~85% (Non Condensing) • Phone Stand • Handset: RJ-7 standard connector IETF drafts supported RFC2327 Session Description Protocol(SDP) RFC3428 SIP Extension for Instance Messaging RFC2976 The SIP INFO Method RFC3515 The SIP Refer Method RFC3261 SIP: Session Initiation Protocol RFC3725 RFC3262 Reliability of Provisional Responses in SIP Best Current Practices for Third Party Call Control in SIP RFC3263 SIP : Locating SIP Servers RFC3842 A Message Summary and Message Waiting Indication Event Package for SIP RFC3264 An Offer/Answer Model with SDP RFC3892 The SIP Referred-By Mechanism RFC3265 SIP – Specific Event Notification RFC3903 SIP Extension for Event State Publication RFC3420 Internet Media Type message/sipfrag RFC4568 SDP Security Descriptions for Media Streams WWW.MOIMSTONE.COM Draft-ietf-sipping-cctransfer-01 SIP call control – Transfer Draft-ietf-sip-replaces-02 The SIP Replaces Header Draft-ietf-sip-sessiontimer-08 The SIP Session Timer Contact E-mail: [email protected] STONEHENGE IP930 /IP950 Conference Butler The best conference solution for conference room ` C Echo Cancellation Self Installing AI Microphone DSP(Digital Signal Processor) Through canceling Static noise and Echo noise, it Regardless of the size of the room, Artificial Intel- For the perfect voice quality, it adapted DSP. It guaran- upgraded the bi-directional sound quality which is most ligence Microphone is automatically installed to tees successfully multilateral conference with clear important function to Conference phone. the best environment of microphone itself. and natural sounds without delaying. 360 omni directional sound recognizing Recording function LCD Display From anywhere in the room Microphone recognizes While doing conference, user can simply record their LCD display shows the telephone number and the the voice. conference through the recording terminal. status of telephone for convenient use. IP Conference Phone PSTN Conference POE (Power over Ethernet) PSTN Conference IP950 earns power from Network Ethernet line. AC/DC IP930 is PSTN conference phone. Connecting to normal circuit, all func- adapter is not necessary. tion for conference phone is available. Speed Dial FXO Port for PSTN Line Speed Dials to save the conference time. IP950 is transferable to PSTN line. User can experience Excellent and flexible scalability from IP950. Wired IP Communication l Wireless IP Communication l Conference Communication l IP Keyphone System www.moimstone.com STONEHENGE IP930/IP950 Specification Call Function • Call Hold • Call Mute • Call Waiting • 3 Party Conference call • Auto Provisioning(HTTPS, HTTP, TFTP) SIP Protocal • Proxy Registration and Failover • Outbound Proxy • Multi-user Registration • Registration Timer • SIP Transport – UDP, TCP, TLS • Secured media negotiation(SRTP) • Realm-based authentication(Digest authentication) • Session Timers • DNS query (A record, SRV, NAPTR) • Codec Negotiation • DTMF relay RTP payload(RFC 2833) or SIP info • Hook flash signaling DTMF/Ring Signal • DTMF (Dual-Tone Multi-Frequency) • Multiple Ring Tones • My bell: 10 Ring • Call Progress Tone Generation (Dial tone, Busy, Audible Ring back) • DTMF generation (RFC 2833 In-Audio or Out-of-band SIP info) Voice & Codec • CNG (Comfort Noise Generation) • Echo Cancellation: G.168 compliance • Codec Auto Negotiation • Codecs - Narrow band -G.711μLaw/aLaw with PLC, G.722.1, G.723.1/A, G.726, G.728, G.729A/AB/E , Broadvoice®16 - Wide band – G.722.1, Broadvoice®32 • Echo Suppression (G.164) • Enhanced Packet Loss Concealment • Silence Suppression (G.164) • VAD (Voice Active Detector) • Adaptive Jitter Buffer • SIPFrag (RFC 3420) • Dynamic Payload Support • Adjustable Audio Frames per Packet • Flexible Dial Plan Support with Inter-Digit Timers Audio Display • Character LCD: 12 character • Language Support: English Power • Microphones: 300 to 3500Hz • Microphone pickup range: up to 3m • Microphone with intelligent microphone mixing • Extension microphones 300 to 3500Hz(Optional) • Dynamic noise reduction IP Network • IPv4 (RFC 791), IPv6(RFC1883) dual stack (optional), TCP, UDP, HTTP, ARP, ICMP • RTP/RTCP, Secure RTP • DNS: A record (RFC 1706), SRV record (RFC 2783) • NAT/PAT • VLAN: IEEE 802.1q • QoS: IEEE 802.1p, DiffServ(RFC 2475) • Network Address Assignment: Static IP/DHCP, PPPoE Managemen • Password Protection for Admin mode and User mode • Management Protocol: SNMPv2 (RFC 2782) • Auto Provisioning: DHCP TFTP, Static TFTP, HTTP • Remote software upgrade: http,tftp • Remote Configuration Ethernet • Dual switched 10/100 Based-T through RJ-45 interfaces • 1 RJ-45 port to connect LAN or Wan, this port supports Defaulted PoE (IEEE802.3af or In line power) • 1 RJ-45 port to connect PC FXO • PoE (Power over Ethernet) defaulted • Power Consumption: 5Watts(max) Technical Spec • Measurements: 310x310x60mm(WxHxL) • Weight: 760g • Color: Dark gray • Operating Temp: 0~45°C • Storage Temp: -20~60°C • Humidity: 10~85% (Non Condensing) 930(PSTN) Specipication PSTN Conference Phone • DSP Chip • Speed Dial button • LCD Display • Measurements : 310x310x60mm(WxHxL) • Weight: 760g • Color: Dark gray • Operating Temp : 0~45°C • Storage Temp : -20~60°C • Humidity: 10~85% (Non Condensing) • 1 RJ-11 FXO port for PSTN back up (Q1,2008) Key & Button • Dial Keys: 12 keys (ITU E.161) • 4 Function Keys: On-hook/Off-hook, Hold, Mute • 2 navigation keys: Up&Down keys can be used for volume control RFCs supported IETF drafts supported RFC2327 Session Description Protocol(SDP) RFC3428 SIP Extension for Instance Messaging RFC2976 The SIP INFO Method RFC3515 The SIP Refer Method RFC3261 SIP: Session Initiation Protocol RFC3725 RFC3262 Reliability of Provisional Responses in SIP Best Current Practices for Third Party Call Control in SIP RFC3263 SIP : Locating SIP Servers RFC3842 A Message Summary and Message Waiting Indication Event Package for SIP RFC3264 An Offer/Answer Model with SDP RFC3892 The SIP Referred-By Mechanism RFC3265 SIP – Specific Event Notification RFC3903 SIP Extension for Event State Publication RFC3420 Internet Media Type message/sipfrag RFC4568 SDP Security Descriptions for Media Streams WWW.MOIMSTONE.COM Draft-ietf-sipping-cctransfer-01 SIP call control – Transfer Draft-ietf-sip-replaces-02 The SIP Replaces Header Draft-ietf-sip-sessiontimer-08 Contact The SIP Session Timer E-mail: [email protected] STONEHENGE IP950 Conference Butler The best conference solution for conference room ` C Echo Cancellation Self Installing AI Microphone DSP(Digital Signal Processor) Through canceling Static noise and Echo noise, it Regardless of the size of the room, Artificial Intel- For the perfect voice quality, it adapted DSP. It guaran- upgraded the bi-directional sound quality which is most ligence Microphone is automatically installed to tees successfully multilateral conference with clear important function to Conference phone. the best environment of microphone itself. and natural sounds without delaying. 360 omni directional sound recognizing Recording function LCD Display From anywhere in the room Microphone recognizes While doing conference, user can simply record their LCD display shows the telephone number and the the voice. conference through the recording terminal. status of telephone for convenient use. IP Conference Phone POE (Power over Ethernet) IP950 earns power from Network Ethernet line. AC/DC adapter is not necessary. FXO Port for PSTN Line IP950 is transferable to PSTN line. User can experience Excellent and flexible scalability from IP950. Wired IP Communication l Wireless IP Communication l Conference Communication l IP Keyphone System www.moimstone.com STONEHENGE IP950 Specification Call Function • Call Hold • Call Mute • Call Waiting • 3 Party Conference call • Auto Provisioning(HTTPS, HTTP, TFTP) SIP Protocal • Proxy Registration and Failover • Outbound Proxy • Multi-user Registration • Registration Timer • SIP Transport – UDP, TCP, TLS • Secured media negotiation(SRTP) • Realm-based authentication(Digest authentication) • Session Timers • DNS query (A record, SRV, NAPTR) • Codec Negotiation • DTMF relay RTP payload(RFC 2833) or SIP info • Hook flash signaling DTMF/Ring Signal • DTMF (Dual-Tone Multi-Frequency) • Multiple Ring Tones • My bell: 10 Ring • Call Progress Tone Generation (Dial tone, Busy, Audible Ring back) • DTMF generation (RFC 2833 In-Audio or Out-of-band SIP info) Voice & Codec • CNG (Comfort Noise Generation) • Echo Cancellation: G.168 compliance • Codec Auto Negotiation • Codecs - Narrow band -G.711μLaw/aLaw with PLC, G.722.1, G.723.1/A, G.726, G.728, G.729A/AB/E , Broadvoice®16 - Wide band – G.722.1, Broadvoice®32 • Echo Suppression (G.164) • Enhanced Packet Loss Concealment • Silence Suppression (G.164) • VAD (Voice Active Detector) • Adaptive Jitter Buffer • SIPFrag (RFC 3420) • Dynamic Payload Support • Adjustable Audio Frames per Packet • Flexible Dial Plan Support with Inter-Digit Timers Audio Display • Character LCD: 12 character • Language Support: English Power • Microphones: 300 to 3500Hz • Microphone pickup range: up to 3m • Microphone with intelligent microphone mixing • Extension microphones 300 to 3500Hz(Optional) • Dynamic noise reduction IP Network • IPv4 (RFC 791), IPv6(RFC1883) dual stack (optional), TCP, UDP, HTTP, ARP, ICMP • RTP/RTCP, Secure RTP • DNS: A record (RFC 1706), SRV record (RFC 2783) • NAT/PAT • VLAN: IEEE 802.1q • QoS: IEEE 802.1p, DiffServ(RFC 2475) • Network Address Assignment: Static IP/DHCP, PPPoE • PoE (Power over Ethernet) defaulted • Power Consumption: 5Watts(max) Technical Spec • Measurements: 310x310x60mm(WxHxL) • Weight: 760g • Color: Dark gray • Operating Temp: 0~45°C • Storage Temp: -20~60°C • Humidity: 10~85% (Non Condensing) Managemen • Password Protection for Admin mode and User mode • Management Protocol: SNMPv2 (RFC 2782) • Auto Provisioning: DHCP TFTP, Static TFTP, HTTP • Remote software upgrade: http,tftp • Remote Configuration Ethernet • Dual switched 10/100 Based-T through RJ-45 interfaces • 1 RJ-45 port to connect LAN or Wan, this port supports Defaulted PoE (IEEE802.3af or In line power) • 1 RJ-45 port to connect PC FXO • 1 RJ-11 FXO port for PSTN back up (Q1,2008) Key & Button • Dial Keys: 12 keys (ITU E.161) • 4 Function Keys: On-hook/Off-hook, Hold, Mute • 2 navigation keys: Up&Down keys can be used for volume control RFCs supported IETF drafts supported RFC2327 Session Description Protocol(SDP) RFC3428 SIP Extension for Instance Messaging RFC2976 The SIP INFO Method RFC3515 The SIP Refer Method RFC3261 SIP: Session Initiation Protocol RFC3725 RFC3262 Reliability of Provisional Responses in SIP Best Current Practices for Third Party Call Control in SIP RFC3263 SIP : Locating SIP Servers RFC3842 A Message Summary and Message Waiting Indication Event Package for SIP RFC3264 An Offer/Answer Model with SDP RFC3892 The SIP Referred-By Mechanism RFC3265 SIP – Specific Event Notification RFC3903 SIP Extension for Event State Publication RFC3420 Internet Media Type message/sipfrag RFC4568 SDP Security Descriptions for Media Streams WWW.MOIMSTONE.COM Draft-ietf-sipping-cctransfer-01 SIP call control – Transfer Draft-ietf-sip-replaces-02 The SIP Replaces Header Draft-ietf-sip-sessiontimer-08 Contact The SIP Session Timer E-mail: [email protected] STONEHENGE IP210 Cost Effective VoIP Phone Great features and Refined design for user Economical VoIP solution for Soho market Business based on standard SIP Optimum solution for various businesses As VoIP phone for business based on standard SIP, it meets every function The best features as contrasted with price. IP210 is the revolutionary and of various IP PBX and soft-switches embodying linked function enough. It economical solution for various business and home user. proved their stability and VoIP technique by installing them into domestic and overseas sites. Based on the proved technique, it increase the convenience and productivity. High Quality sound at conference phone level 4 speed dial and 15 functional buttons 4 speed dials C Speaker phone function built in phone guarantees high quality sound, supporting Full Duplex like conference phone. 5 menu buttons 8 functional buttons Remote upgrading for Effective management SRTP and TLS security Cutting the hacking and wiretapping by SRTP(Secured Real By Remote Management and installation over Web Interface, it makes effective management be possible. time Transport protocol), TLS(Transfer Layer Security), the system supports secured telecommunications. IP Wired Communication l IP Wireless Communication l Conference Communication l IP Keyphone System www.moimstone.com STONEHENGE IP210 Specification Call Function Voice & Codec Key & Button • Call Forwarding • Call Hold • Call screen/ Do Not Disturb • Call Pickup • Call Transfer • Call Waiting • 3-Way Conference call • SIMPLE Based Instant Messaging and Presence • Auto Provisioning (HTTPS, HTTP, TFTP) • CNG (Comfort Noise Generation) • Echo Cancellation : G.168 Compliance • Codec Auto Negotiation • CoDecs - Narrow Band -G.711μLaw/aLaw with PLC, G.722.1, G.723.1/A, G.726, G.728, G.729A/AB/E , Broadvoice®16 - Wide Band – G.722.1, Broadvoice®32 • Echo Suppression (G.164) • Enhanced Packet Loss Concealment • Silence Suppression (G.164) • VAD (Voice Active Detector) • Adaptive Jitter Buffer • SIPFrag (RFC 3420) • Dynamic Payload Support • Adjustable Audio Frames per Packet • Flexible Dial Plan Support with Inter-Digit Timers • Dial Keys : 12 Keys (0 to 9, * and # Key – ITU(E.161) • 8 Function Keys: Redial, Transfer, 3 Way Call, Call Pick Up, Hold, Mute, DND, Call Forwarding • Speaker Phone Button for Conference Call • 4 Memory Dial Keys • 7 Navigating Keys : Up, Down, Left, Right, Menu, Enter, ESC • Volume Up/Down Keys for Ringer, Speaker and Handset Receiver • Phonebook: Record 100 Entries • Call History Logs: Records 60 Entries for In/Outbound Calls and Missed Calls. SIP Protocal • Proxy Registration and Failover • Outbound Proxy • Multi-User Registration • Registration Timer • SIP Transport – UDP, TCP, TLS • Secured Media Negotiation(SRTP) • Realm-Based Authentication(Digest Authentication) • Session Timers • DNS Query (A Record, SRV, NAPTR) • CoDec Negotiation • DTMF Relay RTP Payload(RFC 2833) or SIP Info • Hook Flash Signaling • Visual Message Waiting Indicator DTMF/Ring Signal • DTMF (Dual-Tone Multi-Frequency) • Multiple Ring Tones • My Bell : 10 Ring Tones • Call Progress Tone Generation (Dial Tone, Busy, Audible Ring Back) • DTMF Generation (RFC 2833 In-Audio or Out-of-Band SIP Info) IP Network • IPv4 (RFC 791), IPv6(RFC1883) Dual Stack (Optional), TCP, UDP, HTTP, ARP, ICMP • RTP/RTCP, Secure RTP • DNS: A Record (RFC 1706), SRV Record (RFC 2783) • NAT/PAT • VLAN: IEEE 802.1q • QoS : IEEE 802.1p, DiffServ(RFC 2475) • Network Address Assignment: Static IP/DHCP, PPPoE Managemen Display • LCD : 128x32 Black/White Graphic LCD • Multi-Language Support : English, Korean, Czech, and Others Power • Power Adaptor: Output DC5V/1A, Input AC100~240V(50/60Hz) • Power Consumption: 5Watts (Max) Technical Spec • Dimensions (Main body) : 220x160x61mm (WxHxL) • Weight : 600g • Color : Black / Beige • Operating Temp : 0~45 ° c • Storage Temp : -20~60 °c • Humidity : 10~85% (Non-Condensing) • Password Protection for Admin Mode and User Mode • Management Protocol : SNMPv2 (RFC 2782) • Auto Provisioning : DHCP TFTP, Static TFTP, HTTP • Remote Software Upgrade : http, tftp • Remote Configuration Ethernet • Dual Switched 10/100 Based-T Through RJ-45 Interfaces • 10/100BASE-T : 2 Ports Providing Auto-MDI/MDIX, Enabling the Use of Straight or Crossover Cable in Either Port • 1 RJ-45 Port to Connect LAN or Wan, This Port Supports Optional PoE (IEEE802.3af or In Line Power) • 1 RJ-45 Port to Connect PC RFCs supported IETF drafts supported RFC2327 Session Description Protocol(SDP) RFC3428 SIP Extension for Instance Messaging Draft-ietf-sipping-cc-transfer-01 SIP call control – Transfer RFC2976 The SIP INFO Method RFC3515 The SIP Refer Method Draft-ietf-sip-replaces-02 The SIP Replaces Header RFC3261 SIP: Session Initiation Protocol RFC3725 Draft-ietf-sip-session-timer-08 The SIP Session Timer RFC3262 Reliability of Provisional Responses in SIP Best Current Practices for Third Party Call Control in SIP RFC3263 SIP : Locating SIP Servers RFC3842 A Message Summary and Message Waiting Indication Event Package for SIP RFC3264 An Offer/Answer Model with SDP RFC3892 The SIP Referred-By Mechanism RFC3265 SIP – Specific Event Notification RFC3903 SIP Extension for Event State Publication RFC3420 Internet Media Type message/sipfrag RFC4568 SDP Security Descriptions for Media Streams WWW.MOIMSTONE.COM Contact E-mail: [email protected] STONEHENGE IX200 IP Key Telephone System 새로운 개념의 IP 키폰 시스템 / 중소기업을 위한 차세대 교환기 IP 키폰 시스템을 사용하면… Voice와 Data 통신을 하나의 회선으로 통합하여 더 이상 따로 관리하실 필요 없으며 통합된 회선을 통해 기존 일반전화 서비스는 물론 다양한 VoIP서비스까지 함께 사용이 가능합니다. IP키폰 시스템을 통해 지금까지 사용하시던 전화 시스템보다 훨씬 더 높은 수준의 다양한 기능을 경험하실 수 있습니다. 임베디드 Asterisk 기반의 IP-PBX 성능과 안정성이 이미 입증된 Asterisk 교환기 소프트웨어가 탑재되어 있어 사용자 들이 매우 합리적인 비용으로 그 다양한 기능들을 사용할 수 있습니다. Auto attendant (자동응답기능) 회사의 대표 번호로 전화시 음성안내를 통해 해당 부서로 바로 연결하여 불필요한 전화연결 시간을 줄여 줍니다 확장과 유지보수의 편의성 IX200 은 슬라이딩 보드 교환 방식을 채택 하여 사용량 증가에 따른 업그레이드의 편의성을 고려 하였습니다 안정적 운영을 위한 배터리 및 온도 센서 내장 내장된 백업 배터리는 갑작스런 정전 상황에서도 시스템을 안전하게 보호하고, 시스템 내부 온도를 측정하는 온도 센서는 시스템이 허용 온도를 초과하지 않도록 하여 항상 안정적인 시스템 운영이 가능하도록 설계되어 있습니다. 최대 100명의 사용자 및 20 동시 통화 지원 FXO 12회선까지 수용가능 IX200은 제품은 중소규모 사업자용 IP 키폰 시스템 으로 최대 100명의 사용자를 지원하며, 최대 20개 의 동시 통화가 가능합니다. 2포트의 FXO카드와 4포트의 FXO카드를 선택사양으로 제공하여, 최대 12포트의 FXO회선까지 사용자가 선택하여 확장할 수 있습니다. IP Wired Communication l IP Wireless Communication l Conference Communication l IP Keyphone System www.moimstone.com STONEHENGE IX200 Specification 기본 호 관련 기능 음성 및 코덱 IP 네트워크 • Call Forward - on Busy - on No Answer - Always • Call Blocking • Call Waiting • Incoming/Outgoing Call screening • Call Transfer - Blind Transfer - Attended Transfer • Call Pick-up • Distinctive ringing /Group ringing • Music on Hold • Music on Transfer • Do Not Disturb • Call Back • Call Hold • Conference • Speed Dial • Redial • Message Waiting Indication • CNG (Comfort Noise Generation) • Echo cancellation (G.168, G.165) • Codec Auto Negotiation • Codecs - Narrow band - G.711µLaw/aLaw with PLC, G.723.1/A (pass through), G.726, G.729 A/AB/E , Broadvoice®16 - Wide band - G.722.1, Broadvoice®32 • Echo Suppression (G.164) • VAD(Voice Active Detector) • Adaptive Jitter Buffer • Dynamic Payload Support • Adjustable Audio Frames per Packet • IPv4/IPv6 Dual Stack • SNTP for Clock Synchronization • IEEE802.1q for VLAN • IEEE802.1p, Diffserv for QoS • SNTP(Simple Network Time Protocol) • Static Network Address Assignment • SMTP PBX 기능 • Hunt Groups • Conference Bridging and Access Control • Voicemail System • Call Detail Records • Auto Attendant • Intercom • Find me/Follow me • Automatic/Manual redial • Call Parking and Call Retrieval • Caller ID • Wake-up Call • Interactive Voice Response • Linear or Random Music Play • Web Interface for Voicemail - Checking - E-mail notification of Voicemail - Voice Message Forwarding • PC client programs for - SMS - IM & Presence - Click to dial - Phonebook (User/Group) • Flexible Dial Plan with Inter-Digit Timers • Versatile Management Features • Web Based system configurations • Remote diagnostics and software upgrade SIP 프로토콜 • Proxy Registration and Failover • Outbound Proxy • Multi-user Registration • Registration Timer • SIP Transport – UDP, TCP, TLS • Secured Media Negotiation (STRP) • Realm-Based Authentication(Digest authentication) • Session Timers • DNS query (A record, SRV, NAPTR) • Codec Negotiation • DTMF relay RTP payload (RFC2833) or SIP info • Hook Flash Signaling • Visual Message Waiting Indicator CPU 보드 • 400MHz Mips CPU • 1GB Flash Memory • 1 Ethernet 10/100base TX port (RJ45) • Reset and Factory Default Button • 6 LEDs for System Status Indication TA 보드 • 2 or 4 FXO ports to the Central Office (RJ11) • 5 LEDs for board status indication • Loop start Line interface • Caller ID support 전원 어댑터 • Input 100 ~ 240VAC, 50~60Hz, 1.2A • Output 12.0Vdc, 1.5A 기타 기술 사양 • Dimension: 291x221x55 (WxHxL) • Weight: 4Kg • Humidity: 5% ~ 90% Non-condensing • Operating Temperature: 0 °C ~ 50 °C IX200 Model Code Model Code Description IX200-XBS-01A IX200 IP PBX Base Station(/w Batter, Back Plane, FAN) IX200-PCD-01A Main Power Cord, 2.5m, Korean Style IX200-ECA-01A Ethernet Cable, Cat 5, 20m IX200-DPA-01A DC Power Adaptor IX200-APU-01A IX200 IP Key System Host IX200-APM-01A IX200 IP Key System Host Connector Board IX200-DTA-01A IX200 Dual Terminal Adaptor Board IX200-DTM-01A IX200 Dual Terminal Adaptor Connector Board IX200-QTA-01A IX200 Quad Terminal Adaptor Board IX200-QTA-01A IX200 Quad Terminal Adaptor Connector Board WWW.MOIMSTONE.COM Main Board FXO Board Contact E-mail: [email protected]