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Istruzioni per l'uso ZyXEL Communications, Modello PRESTIGE P-2002

Produttore : ZyXEL Communications
File Size : 160.54 kb
File Nome : 69416af3-d2ad-e724-b98b-d6bcc0c0f73c.pdf
Lingua di insegnamento: en
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Facilità d'uso

Connecting a traditional phone set to the POTS phone port of the P-2002 series instantly establishes an IP phone device, allowing users to place VoIP calls without the use of a PC. Based on industry standard SIP (Session Initiation protocol, RFC 3261) call control protocol and sophisticated voice compression technology, the P-2002 series offers smooth IP-based voice communication with superb sound quality. ZyXEL’s P-2002 series Analog Telephone Adapter is an ideal solution for IP Telephony service deployment as well as SOHO and Home VoIP applications. A p p l i c a t i o n D i a g r a m VoIP Analog Telephone Adapter Supports Open Standard ZyXEL’s P-2002 series supports standard SIP (RFC 3261), which is suitable for IP telephony service deployment and addresses the needs of users who are connecting to emerging SIP-based communication networks. Easy to Deploy ZyXEL’s P-2002 series supports multiple methods of configuration including Web-based configuration, Telnet management, and TFTP Auto-Provisioning. It eliminates the time and effort associated with complicated installation procedures. Multiple SIP and Voice Channels Based on ZyXEL’s flexible VoIP technology platform, one or more SIP phone number may be configured with the P-2002 ATA. Each SIP phone number can be freely assigned to either phone port individually, or to both ports at the same time. By using the P-2002 ATA to deploy VoIP service, service providers need only provide a minimum of one SIP number to each subscriber. Billing and service deployment are simplified by allowing subscribers to use two phones with the same SIP number. If one telephone is in use, users may use a secondary telephone to place a call while using the same SIP number. Additionally, users may also receive incoming calls even when the SIP number is in use. The advantages of the P- 2002 ATA include increased traffic and service revenue. Cost Effective ZyXEL’s P-2002 series turns a traditional phone into an IP phone device at a very low cost. Not only does the Prestige 2002 series deliver VoIP service at a low per-port cost but the built-in two-port Ethernet switch also allows user to connect their PC to the reserve Ethernet port without adding an extra Ethernet switch or hub. Lifeline Support (Option, P-2002L) ZyXEL’s P-2002L allows users to maintain PSTN phone service and VoIP service at the same time. When ATA power down occurs or VoIP service is not available, outgoing phone calls will be relayed to PSTN automatically. When VoIP service is active, users can place calls to a PSTN phone by dialing pre-fix numbers. Emergency calls or important calls can be relayed to PSTN automatically. IP Telephony Service Provider Application Trunking Gateway VoIP Call Server IP Telephony Service Provider PSTN Phone Cable/DSL Modem PSTN Network PSTN Network Home Cable/DSL Modem P-2002 FXS FXS Home FXS FXS P-2002L Internet RJ-45 Ethernet RJ-45 Ethernet Life Line Relay IP-PBX based Corporate VoIP Application IP Network PSTN Phone Headquarters IP-PBX Branch Office FXS FXS P-2002 P-2002 Traditional Phone Network RJ-45 Ethernet RJ-45 Ethernet PSTN Network Prestige 2002 Series V oIP Analog T elephone Adapter Prestige 2002 SeriesVoIP Analog Telephone Adapter Features Voice Functionality • Fax Tone Detection and Pass-through PSTN Lifeline Support • SIP (RFC 3261) version 2 • Modem Tone Detection and Pass-through* (Option, P-2002L with PSTN Line Port) • SDP (RFC 2327) • T.38 FAX Relay* • Send Emergency Call to PSTN • RTP (RFC1889) • Speed Dial Phonebook • Make a PSTN Call by Dialing • RTCP (RFC1890) • Telephony Features: Caller ID, Call Pre-fix Number • Echo Cancellation: G.168 Waiting, Message Waiting, Three Way • PSTN Relay when Device Power Down • VAD (Voice Activity Detection) Conference (Depends on IP Telephony • PSTN Relay when VoIP Service is • Silence Suppression Service Provider)* Not Available • CNG (Comfort Noise Generation) • Flash Hook Timer* • Receive Incoming PSTN Phone Call • Dynamic Jitter Buffer (Adaptive) • Multiple SIP Accounts/Phone Numbers — Network Functionality • QoS Supports TOS, DiffServ Freely assignable to Each Phone Port • PPPoE • VLAN Tagging • Multiple Channel Technology • DHCP Client • 3 REN per RJ-11 FXS ports Voice Codecs • Friendly Web-based Configuration Tool • DTMF Detection and Generation • G.711 • Telnet Management • DTMF: In-Band and Out-of-Band • G.729 • FTP/TFTP Firmware Upgrade and (RFC 2833), (SIP INFO) • GSM (Option)* Configuration Backup/restore • SIP NAT Traversal Support STUN • G.726 (Option)* • Support TFTP Auto-Provisioning (RFC 3489) • G.723 (Option)* Specifications Hardware Specification • Reset Button Physical Specification • LAN: Two 10M/100M Auto MDI/MDIX • Status LEDs Indicator: PWR/VoIP, LAN, • Dimensions: Ethernet Ports PC, Phone1, Phone2 180(L) x 128(D) x 36.5(H) mm • Phone: Two RJ-11 ports for Analog • Power: 12V AC input • Weight: 312g Telephone sevice Operating Environment • Lifeline Port: One RJ-11 Port for PSTN • Temperature: 0°C ~ 40°C L...

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