Patton Smartnode 4112 VoIP Gateway 2 FXO Ports

Patton Smartnode 4112 VoIP Gateway 2 FXO Ports
[SN4112/JO/EUI]

£298.80 ex VAT
£358.56 inc VAT

Patton Smartnode 4112 VoIP Gateway 2 FXO Ports

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Availability: In Stock

Patton Smartnode 4112 VoIP Gateway 2 FXO Ports

The business-class SmartNode 4112 VoIP Media Gateway supports up to two transparent phone calls and leverages VoIP for carrier and corporate access. Connecting to any analog phone, fax, or PBX, the SN4112 is an effective and flexible solution for toll-bypass, remote/branch office voice connectivity, and enhanced carrier services.

The SN4110 series is the perfect choice for phone-to-IP connectivity supporting up to 8 FXS ports or a combination of 4 FXS and 2 or 4 FXO ports. With its FXS analog ports the SN4110 connects to any legacy telephone or PBX and provides dial-tone, ringing, caller-ID and other services. When equipped with FXO ports, the local PSTN can be accessed enabling local calling and enhanced toll-bypass applications while using a single connected telephone. Flexible call integration allows per-port telephone numbers, programmable call progress tones, and distinctive ringing. With Telephony-over-IP (ToIP) call switching, calls can automatically be routed to the PSTN or the IP network while providing flexible numbering plans and end-to-end feature transparency. PPPoE, DHCP, and VLAN offers universal IP connectivity and optional IPSEC VPN with AES/3DES guarantees secure voice over the public network.

Patton's SmartNode 4110 delivers the legacy phone interfaces, service transparency, and flexible PSTN integration for true converged packet voice.

Voice Connectivity

2-wire loop-start presented as an RJ-11/12
2.5kV line isolation Surge Protection: Voice Ports: Tip & Ring protected by 270 V side actor
Off-hook and ring detection, Automatic line gain, Programmable ring count
End of Call detection, Line drop, busy tone, battery reversal detection
Hook-Flash Sending, Hook-Flash relay, DTMF send, detect, and relay
Caller ID FSK CLI reception and relay (Bellcore/ANSI and ETSI/ITU), Call routing based on Caller ID
Second dial-tone for two-stage DTMF dialing, Call routing based on DTMF numbers

Data Connectivity

10/100 Full Duplex, Autosensing, Ethernet RJ-45 port

Voice Processing (signalling dependent)

Voice codes
- G.711 A-Law/µ-Law (64kbps)
- G.726 (ADPCM 40, 32, 24, 16 kpbs)
- G.723.1 (5.3 or 6.3 kbps)
- G.729ab (8kbps)
Up to 8 parallel voice connections
G.168 echo cancellation
Carrier tone detection and generation
Silence suppression and comfort noise
Configurable dejitter buffer
Configurable tones (dial, ringing, busy)
RTP/RTCP (RFC 1889)

Fax and Modem Support

G.711 Fax- and Modem-Bypass

T.38 Fax relay (9.6 k, 14.4 k)

Voice Signalling

H.323v4
- RAS, H.225, H.245
- H.235 secure RAS
- Fast-connect, early H.245
- Gatekeeper autodiscovery, Alias registration
- Overlap dialing
- Empty capability set (call transfer, hold)
- H.323v1 call transfer, hold
SIPv2
- Supported RFCs: 2806, 2833, 2976, 3261, 3263, 3264, 3323, 3325, 3515
- draft-ietf-sip-cc-transfer-02, draft-ietf-sip-cc-transfer-05
- draft-ietf-sip-refer-02, draft-ietf-sip-replaces-01
- draft-ietf-sip-session-timer-04, draft-ietf-sip-session-timer-08
- Caller ID, CLIR (receive from PSTN)
- Support for proxy and redirect servers
- RFC2833: DTMF Relay, SIP INFO DTMF Relay, Fax-Bypass (G.711), T.38 Fax-Relay
- Session Timer, Record-Routing, Authentication
- Compression CODECs, Fax/Modem bypass

Call Routing

Virtual Interfaces
Routing Criteria: Called party number (Destination) Calling party number (Source) Time of day, day of week, date
Longest prefix match, wildcard match, regular expression match
Number Manipulation Functions: Replace numbers Add/remove digits Regular Expressions
Fallback Routing: Soft Fallback to alternative interface or Call Router table

IP Services