VoIP Gateway(SVR404 SVR400 SVR402 )

Product Overview The ALCON SVR400 Series, a Hybrid SIP Proxy Server, registers and authenticates users, and routes calls between user agents. With SVR400 Series SIP Server, you can use SIP Agents, SIP soft phones, and SIP Gateways for VoIP communications. The SVR400 Series also provide Trunk Route capability to call Least Cost Route to ITSP or PSTN line. Otherwise SVR400 Series were implemented telephone interface connected with PSTN, PBX or telephone set for SIP Communication.Select the least cost call FeatureFor example, you can send all international calls through 4 different ITSP by the least call rate. The feature can select the most effective service provider by call route rule setting. Hybrid SIP Proxy Server Key Features 1. Least Cost Trunk Route : SVR 400 series provide 4 Trunk Route Setting for Off-Net call to ITSP or Termination Gateway. (SVR series can subscribe to most 4 Telephone Service Provider).2. 100 SIP endpoints registration / 250 SIP accounts scale: SVR400 series provide 100 SIP end devices to register and 250 SIP accounts management. 3. Register Security Policy: SVR400 series providing MD5 authentication setting.4. Real time Call Detail Record and Post Call Detail Record Report : Support Real Time CDR to monitor VoIP calls, including caller ip, called ip , call date , call duration and other information. Also providing a CDR report to look up VoIP call record.5 .Top 20 list: SVR 400 series can lists top 20 calls by call duration, caller number, calling number, caller IP or called IP address.6. Syslog client: Send CDR information to Syslog server. Gateway Key Features 1. Caller ID Delivery: FXS support DTMF&FSK Caller ID generation.2. Smart VoIP call Dialing Book: VoIP call Book could provide any application VoIP call to any type destination (Domain name/IP address, PSTN or PBX) or hunting number setting.3. AC termination Impedance: 600/900 OHM and complex impedance.4. Polarity Reversal Detection: Type I and Type II.5. Smart-QoS Guaranteed: This bandwidth management feature provide good voice quality when user place VoIP call and access internet at the same time. The gateway will start reserve bandwidth for voice traffic automatically when VoIP call proceeds.6. Voice channels status display: This function display each port status like as on-hook, off-hook,? calling number called number, talk duration, codec. Gateway Telephony Specification 1. Voice Codec: G.711(A-law /?-law), G.729 AB, G.723 (6.3 Kbps / 5.3Kbps) 2. FAX support : T.30 / T.383. Echo Cancellation: G.165/G1684. FXO hang up detection / anti-seized : Tone Learning Automatically / Manual Tone Learning.5. Answer supervision: Support Battery Reverse Detection and Voice Detection.6. FXO answer delay time: Support delay 0 8000 ms to answer.7. Adjustable AC Termination Impedance : 600 / 900 OHM and complex Impedance8. FXO answer Mode: FXO provides Ringing Answer, Connection Answer and Non Answer for? configuration. IP Specifications 1. SIP (RFC 3261), SDP (RFC 2327), RTP Payload for DTMF Digits (RFC2833).2. LAN: NAT, Virtual Server, DHCP Server3. WAN: PPPoE client, DHCP client, Fix IP Address, DDNS client4. Network Address Translation: Providing build-in NAT router function.5. Smart QoS: Guarantee the voice bandwidth Call Features 1. Voice channels status display 2. Direct Dialing Mode : peer to peer call3. Adjustable volume : - 9 db ~ 9 db4. Silence Compression5. Auto Dial for speed6. Dynamic Jitter Buffer Configuration & Management 1. Web-based graphical user interface 2. Remote management over the IP network3. FTP firmware upgrade4. Backup and Restore Configuration file General Specification 1. AC power : AC100V-240V, DC12V/1.5A,50/60 Hz 2. Temperature: 0