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Shang308 FXO and FXS Gateway Intelligent Call Routing Gateway Solution for VoIP
Toll Quality Voice and Carrier-Grade Feature Support The Shang308 delivers clear, high-quality voice communication in diverse network conditions. Excellent voice quality in a demanding IP network is consistently achieved via our advanced implementation of standard voice coding algorithms. The Shang308 is interoperable with common telephony equipment like voicemail, Fax, PBX, and interactive voice response systems. It use the same software with Shang332, so the specification is same to Shang332 Powerful number manipulation Number manipulation make the calls route between multi lines more smoothly. Telephony • Service Authentication via Caller ID • Per Call Authentication and Associated Routing • Least Cost Routing Support • Impedance Agnostics • Call Waiting, Cancel Call Waiting, Call Waiting Caller ID Detection (Bell core, ETSI) • Caller ID with Name/Number (Multi-national Variants) • Caller ID Blocking • Call Forwarding to PSTN or VoIP Service: No answer, Busy, All • Do Not Disturb • Call Transfer • Call Blocking with Toll Restriction • Delayed Disconnect • Off-hook Warning Tone • Selective/Anonymous Call Rejection • Hot line and Warm Line Calling • Speed Dialing • Fax: G.711 Pass Through or Real Time Fax over IP via T.38 Product Specific • VoIP to PSTN Service Call Origination and Termination • PSTN to VoIP Service Call Origination and Termination • Single Stage and Two Stage Dialing • Forward Calls to VoIP service • Forward Calls to PSTN service • PSTN Line Sharing with Multiple Extensions • Automatic PSTN Fallback (Loss of Power or IP Service to Unit - with Quiescence to Normal Operations) • Advanced Inbound and Outbound Call Routing • Independent Configurable Dial Plans - Up to 50 • Force PSTN Disconnection • Sequential Dialing Support VoIP to PSTN Authentication and Routing Features • VoIP to PSTN Gateway Enable/Disable • One Stage Dialing Enable/Disable • VoIP Caller ID Pattern Matching • VoIP Access Allowed Caller List (No Further Authentication) PSTN to VoIP Authentication and Features • PSTN to VoIP Gateway Enable/Disable • Ring Through to FXS Enable/Disable • Ring Tone - Configurable • Caller ID (Bellcore ,ETSI) for VoIP Service Access • Caller ID Enable/Disable • Access Allowed Caller List (No Further Authentication) • Least Cost Routing (via Outbound VoIP - Line1 Dial Plan) FXO Behavior Features • VoIP Answer Delay Timer • PSTN Answer Delay Timer • PSTN Ring Through Delay Timer • PSTN Dialing Delay Timer PSTN Disconnection Detection Features • CPC (Removal of Tip/Ring Voltage Momentarily) • Polarity Reversal • Long Silence (Configurable Time Setting) • Disconnect Tone (e.g. Reorder Tone) • Silence Threshold International Control Features • FXO Port Impedance - Configurable to 16 settings • Ring Frequency - Configurable • PSTN Gain Settings • Ring Frequency - Maximum Setting • Ring Validation Time Setting • Tip/Ring Voltage Adjustment Setting • Ring Indication Delay Setting • Operational Loop Current Minimum Value • Ring Time-out Setting • On-Hook Speed Settgwing • Ringer Impedance Setting • Line-in-Use Voltage Setting Specifications MAC Address (IEEE 802.3) IPv4 - Internet Protocol v4 (RFC 791) upgradeable to v6 (RFC 1883) ARP - Address Resolution Protocol DNS - A Record (RFC 1706), SRV Record (RFC 2782) DHCP Client - Dynamic Host Configuration Protocol (RFC 2131) DHCP Server - Dynamic Host Configuration Protocol (RFC 2131) PPoE Client - Point to Point Protocol over Ethernet (RFC 2516) ICMP - Internet Control Message Protocol (RFC792) TCP - Transmission Control Protocol (RFC793) UDP - User Datagram Protocol (RFC768) RTP - Real Time Protocol (RFC 1889) (RFC 1890) RTCP - Real Time Control Protocol (RFC 1889) SNTP - Simple Network Time Protocol (RFC 2030) QoS - Voice Packet Prioritization over Other Packet Types Router or Bridge Mode of Operation MAC Address Cloning Port Forwarding SIPv2: Session Initiation Protocol v2 (RFC 3261, 3262, 3263, 3264) SIP Proxy Redundancy - Dynamic via DNS SRV, A Records Re-registration with Primary SIP Proxy Server SIP Support in Network Address Translation Networks - NAT G.711 (A-law and μ-law) G.726 (16/24/32/40 kbps) G.729 A G.723.1 (6.3 kbps, 5.3 kbps) Dynamic Payload Adjustable Audio Frames per Packet Fax Tone Detection and Pass-Through (Using G.711) Fax Pass-Though - Using G.711 DTMF: In-band & Out-of-band (RFC 2833) (SIP Info) Flexible Dial Plan Support with Interdigit Timers and IP Dialing Call Progress Tone Generation Jitter Buffer - Adaptive Frame Loss Concealment Full Duplex Audio Echo Cancellation (G.165/G.168) VAD - Voice Activity Detection with Silence Suppression Attenuation / Gain Adjustments Flash Hook Timer Polarity Control Hook Flash Event Signaling Caller ID Generation (Name & Number) - Bellcore, DTMF, ETSI System Reset to Factory Default Password Protected Admin and User Access Authority Provisioning/Configuration/Authentication: Web Browser Administration & Configuration via Integrated Web Server Telephone Key Pad Configuration with Interactive Voice Prompts Automated Provisioning & Upgrade via HTTP, TFTP Non-intrusive, In-Service Upgrades Report Generation & Event Logging 2 100baseT RJ-45 Ethernet Port (IEEE 802.3) -- 1 WAN, 1 LAN 32 RJ-11 FXS maximum Phone Ports - For Analog Circuit Telephone Device (Tip/Ring) 32 RJ-11 FXO maximum Phone Ports - For a Telco or PBX Connection Ring Waveform: Trapezoidal and Sinusoidal Terminating Impedance: Configurable Settings including North America 600 ohms, European CTR21 FCC , CE, Administration Guide - Service Providers Only Provisioning Guide - Service Providers Only 32ºF to 113º F (0ºC to 45º C) -13ºF to 185º F (-25ºC to 85º C) 10 to 90% Non-condensing, operating and non-operating |
