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Patton SmartNode 4110 Series

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Patton SmartNode 4110 Series

Patton SmartNode 4110 Series

Patton SmartNode 4110 Series


Patton SmartNode 4110 Series
Patton SmartNode 4110 Series

Analog VoIP Gateway with up to 8 FXS/FXO ports

The SmartNode 4110 VoIP Media Gateway supports up to eight FXS or FXO telephone connections. Connect PSTN Lines, PBXs, and standard phones for voice and fax over any IP network. Now the corporate, small, or remote office can access Internet telephony services, eliminate toll charges and route calls to and from the PSTN, Internet, or LAN.

Features & Benefits

10/100 Full-Duplex Auto-Sensing Auto-MDX Ethernet port
14 LEDs for System, Ethernet, and Call status
Universal 100-240VAC (contact us for 48VDC Power)
Configuration and Management through Webinterface, CLI, Telnet, Console and SNMP
Up to 8 Voice and FAX Calls over IP Support-Use any CODEC or T.38 FAX on any port, any time
2,4,6 or 8 FXS ports connect to your standard telephone with programmable tones, ringing, CLIP etc.
2 or 4 FXO ports connect to a PBX or the PSTN.
Programmable call routing switches between any FXS, FXO and IP connection.
Toll Quality CODECs & T.38 FAX-Use standard G.711 or G.726 Codecs for toll-quality or G.723.1 or G.728ab for low-bandwidth applications. T.38 FAX and FAX Bypass features allow high-quality FoIP.
H.323v4, SIP Signaling-Deploy into any enterprise or carrier softswitch network with the leading call and session signalling protocols. Maximize Interoperability to create advanced multimedia and messaging services.
IPSEC VPN, Encryption, & VLAN-Connect your telephone to IP and secure your calls with optional IPSEC. Choose DES/3DES or AES for encryption. VLAN tagging/stripping allows interfacing into your 802.1pQ network.
Full ToIP™ Call Switching-Enhanced circuit-switched call routing allows user programmable and adaptive call handling based on hunt groups, caller/called ID, and time of day.

The SmartNode 4110 VoIP Media Gateway supports up to eight transparent phone calls while leveraging VoIP for lower-cost carrier and corporate access. Connecting to any analog phone, fax, or PBX, the SN4110 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, and caller-ID. 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 select the least-cost-route 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.

Patton SmartNode 4110 Series

Application-Remote Office/Branch Office Voice Extension and Access
In enterprise networks, transparent access to PBX features while using existing equipment is key to low-cost operations. Now, instead of installing a separate PBX at the remote office, the SmartNode 4110 is able to provide transparent extension while simultaneously connecting multiple locations. The extensions can be managed centrally and benefit from PBX services such as calling groups, least cost routing, and call forwarding. PSTN access allows local calls to be processed without using corporate remote PBX resources. Additionally, the corporate PBX can break-out and bypass any long distance charges by using the remote office for the local gateway.

Specifications

Voice Connectivity o 2, 4, 6, or 8 FXS ports

2-wire Loopstart, RJ-11/12
Short haul loop 1.1 KM @3REN
EuroPOTS (ETSI EG201 188)
Programmable AC impedence, feeding, and ring voltage; On-Hook Voltage 29VDC
Caller-ID Type-1/2 FSK and ITU V.23/Bell 202 generation
2 or 4 FXO ports
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, H.245 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

RFC2806: URLs for Telephone Calls, RFC3261: SIP: Session Initiation Protocol
RFC3263: Session Initiation Protocol (SIP): Locating SIP Servers
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 (without CNIP), CLIR (receive from PSTN)
Support for proxy and redirect servers
RFC2833: 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

IPv4 router
Static Routes, ICMP redirect (RFC 792), RIPv1, v2 (RFC 1058 and 2453)
DiffServe/ToS Packet labeling
802.1p VLAN tagging
Access Control Lists
IPSEC AH & ESP Modes, preshared Keys
AES/DES/3DES Encryption

Management

Web GUI
Industry standard CLI with local console (CRJ-45, RS-232) and remote Telnet access
TFTP configuration & firmware loading
SNMP v1 agent (MIB II and private MIB)
Built-in diagnostic tools (trace, debug)

Operating Environment

Operating temperature: 0 - 40°C

Operating humidity: 5 - 80% (non condensing)

System

CPU Motorola MPC859 @ 50 MHz
Memory 32MB SDRAM/4MB Flash
Power: 100 - 240 VAC (50/60 Hz)
Power dissipation: 4 - 12W model dependent
Compliance

EMC compliance: EN55022 and EN55024
Safety compliance: EN 50950
CE compliance
FCC Part 15 Class A




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