LotusGate 1024PS 24-Line FXS SIP IP Gateway
Contents
1. e Support up to 16 SIP proxy Servers e Support 24 analog Phone sets at one IP address e Support different SIP Trunk to each FXS line e Auto HTTP Provision feature LotusGate 1024PS e Flexible Routes Plan and Dial Plan e Redundant Firmware Image Introduction LotusGate 1024PS is an 24 line FXS gateway with SIP protocol IP device which allows to connect 24 sets of analog telephone to make or receive VoIP call over Internet or VPN network through Internet Telephony service provider This device is suitable for office user through ITSP service provider to install at office or branch office to call between different offices It can be installed at basement of apartment or dweller building to provide analog phone set to each house to make receive telephony internet call via Telco or ITSP s broadband device and network To select up to 16 SIP service Accounts LotusGate 1024PS is appropriate to use up to 16 VoIP Service Providers IP Centrex service and IP PBX within offices and remote branch offices One of 16 SIP Servers or ITSP Service provider or alternative IP PBX can be configured freely at each line FXS port to make or receive IP Call It provides 16 service platforms to select lowest rate or different purpose according to your dial number or country or application 19 inch 1U chassis easy to install LotusGate 1024PS is an 19 inch 1U chassis and suitable to install at Relay Rack for vast lines installation at office Tel2. o T 38 FAX ECM Redundant depth Volume o FAX Relay T 38 or Disable o Retrieve Voice Mail from IP PBX o FXS Caller ID Mode Transparent or Inhibit o SIP Caller ID Mode Transparent or Inhibit Support Peer to Peer Dialing o Flash Time Detection range from 80 to 800 ms o ON HOOK Voltage 48Vdc o Ring Sine Wave frequency 10 to 70 HZ o Ring Level 10 to 95 Vrms o Ring REN 3 o Configure Ring ON time 0 to 8000 ms o Configure Ring OFF time 0 to 8000 ms o Configure Ring Cadence Frequency and Voltage 0 Support Polarity reversal for Billing O Service Up to 1 Kilo meter distance to analog telephone set o Generate Current Drop Time Open Loop Disconnect time e MANAGEMENT o Administrative Telnet CLI and HTTP HTTPS with desired Port number o Enable Disable HTTPS and Telnet Service o http provision through MAC address o RS 232 console cable to configure WAN LAN IP Address o Multilingual Web User Interface o 3 Levels of User Access Right with Password protection Administrator Supervisor and User o HTTP HTTPS Service Access limitation from WAN port o Provides System Status Logs o Status display Network Line SIP Trunk status o Diagnostics debug through Syslog Event Notice o Debug in real time by Telnet o 8 Debug Level Emergency Alert Critical Error Warning Notice Information Debug o Auto Provision via HTTP Server Support SNMP V2 Trap o Configuration Backup Restore o Du
3. al Firmware Image Backup o Reset to factory Default Support LotusCom proprietary encryption protocol at SIP Signal and Voice codec during transmitting to IP network in order to Anti ISP block of VoIP call This feature only be available with LotusCom SIP server or IP PBX3020IPv6 e Environmental o Actual Dimension 44 W x4 4 H x26 2 D CM o 19 inch 1U chassis with Relay Rack Mount Bracket o Weight 4 3kg One unit with packing o Operating Temp amp Humidity Temp 0 C 45 C 32 F 113 F 10 90 relative Humidity humidity non condensing o Power Input AC100V to 240V 50 60Hz e Packing Accessories o LotusGate 1024PS gateway x 1 pcs o Relay Rack Mount Bracket x 2 pcs o AC Power cable x 1 pcs o CD User Manual x 1 pcs e Warranty o One year Y Lotus Communication Inc 459 Dodson Lake Dr Arlington TX USA 76012 Tel 1 817 548 1510 e Mail info lotuscomsys com Y Lotus Communication Inc MEGNA office No 126 DIC16 Dubai Internet City Dubai UAE Tel 971 4 390 0828 Cell 971 50 151 1901 e Mail MEGNA lotuscomsys com URL www lotuscomsys com
4. co and ITSP service provider 1 WAN and 1 LAN with NAT feature together with RS 232 DB 9 local console port allows engineer to configure and maintain this device locally and remotely It can be installed at outdoor electric pole and sidewalk There are 24 ports RJ 11 line connectors to connect to telephone cable via existing Telecom MDF No more new cable is needed Flexible Dial plan and Route Plan Features LotusGate 1024PS provides flexible Dial Plan from FXS to IP Trunk SIP Softswitch Dial Plan is to configure in what condition the digits can be sent out to IP network The dial inter digit time before dialing is configurable to meet busy users or home user Dial Rule is able to detect the prefix code and maximum digits reached and then dial out automatically The Digit Manipulation DM allows you to configure matched prefix code digits length start and stop digit position to be replaced digits as well Routes Plan is to configure the incoming and outgoing call routes which you desire this call to go out or allow to income For instance IP incoming call may Ring to FXS port with Priority Cyclic or Simultaneously ring You can also configure IP incoming call by Matched prefix digits Matched incoming FXS line number and Matched digit length For FXS outgoing call routes the hunting type supports Priority Cyclic or Simultaneously ring and select which SIP trunk SIP Proxy Server to go FXS outgoing call routes also support by Matched p
5. g Back Dial Busy call waiting ROH Warning Holding Stutter dial tone and disconnect tone o Configure Tone Frequency Cadence Level and Cycle o Global Country Based Tone Specification o NAT Traversal support STUN UPnP and Behind NAT Out Band DTMF RFC2833 and SIP Info o Configure DTMF send ON and OFF time o Configure DTMF detect Min ON and OFF time o DTMF Relay Volume o Flash Time transmit via SIP Info Enable or Disable o Message Waiting Indication Stutter Tone Notice Speed Dialing Call Waiting Switching between Calls Call Forward Busy Unconditional No Answer o No Answer Time out o Block Anonymous Call o Hot Line o Call Hold Call Transfer o Flexible Dial Plan o First digit and Inter digit time out timer setup o Manual SEND digits selection key o Retrieve transfer call from 3rd party by dial Code default o Dial Rule Prefix and Maximum digits digit position to replace o Digit Manipulation Drop and Replace Rule o Extension Hunting Support SIP Trunk up to 16 SIP Servers o Import and Export SIP Trunk configuration Outgoing SIP Caller ID Selection o Accept desired SIP Proxy incoming calls Only o Flexible Routing Plan o Prefix Match and Length o Matched FXS Line port number o Incoming call type FXS or VoIP o Priority Ring Cyclic Ring o Simultaneous Ring o Programmable Hunting Cycle o Backup Routing with Digit Manipulation o Default Routing
6. refix digits Matched incoming FXS line number and Matched digit length Both direction supports No Answer time out and Backup Routes Specification o DHCP Client RFC 2131 WAN port e Interface o DHCP Server LAN port o Ethernet port RJ 45 10 100 base T o NAT Server RFC 1631 o 1 WAN port connect to IP Network o PPPoE Client o 1 LAN port connect to PC with NAT o DDNS DynDNS O Support Bridge NAT and Gateway mode o DNS Client o Telephony port RJ 11 x 24 pcs o Firewall o RS 232 Console port DB9 Male o URL Filter o AC power input Jack o IP Filter o AC Power ON OFF Switch o MAC Address Filter o LED Indicator for System SIP and FXS status o Application program Filter o Port Filter e IP Network connection o Port Forwarding o IPv4 RFC 791 Bandwidth Control Download and Upload o MAC Address IEEE 802 3 Maximum Bandwidth and reserved bandwidth o MAC Clone Setting UPnP Server at LAN port Vendor Class ID Behind NAT use DMZ for NAT traversal o IP ICMP ARP RARP SNTP o SNTP with time zone and Daylight Saving Static IP o TCP UDP RFC 793 768 o RTP RTCP RFC 1889 1890 o IPV4 ICMP RFC 792 o TFTP Client ie VLAN Support 802 1Q 802 1P ie VLAN ID Range 2 to 4094 o VLAN Priority 0 to 7 QoS DiffServ RFC 2475 TOS RFC791 1394 SIP Protocol o RFC3261 compliance Support up to 16 SIP Server Register Accounts SIP UDP Protocol O Support SIP compac
7. t Form Support SIP HOLD Type o SIP Session Timer RFC 4028 o Configure SIP port and SIP QoS Type o MDS Digest Authentication RFC2069 RFC2617 O SIP PRACK RFC3262 o Early Delay Media support o Offer Answer RFC3265 ie Message Waiting Indication RFC3842 o Event Notification RFC3265 o REFER RFC3515 Support Outbound Proxy o SIP Proxy Keep Alive time setup Support Primary and Secondary SIP Server O Support STUN NAT Traversal O Support rport parameter RFC 3581 Audio Codec o G 711 A law u law G 729A G 723 1 6 3K 5 3K GSM FR Full Rate 13kbps o Select voice codec priority Local or Remote o Configure RTP port and RTP QoS Type o Silence Suppression o VAD CNG o LEC Line Echo Canceller o Max Echo Tail Length G 168 32 64 and 128ms o Packet Loss Compensation o Input Encode Gain setup o Output Decode Gain setup o In band out of band DTMF RFC4733 RFC2833 SIP INFO o Adaptive Configurable Jitter Buffer o G 168 Acoustic Echo Cancellation o Dialing Plan with drop replace Insert dialing digits o Select First digit and Inter digit timeout dura tion Sec o Selectable Call Progress Tone O Support Specified Line Calling Call Features o Caller ID display DTMF before after 1st ring and FSK before 1st ring ETSI and Bellcore o DTMF Caller ID start and stop BIT A to F configurable o Polarity Reversal before Caller ID or not o Tone Generation Ring Rin
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