Cisco SPA3102
Voice Gateway with Router
SPA3102-NA100% working & tested, like brand new.
(Used for 1 month & now factory default settings are set for the new owners!)
It seems to be factory unlocked, no password is required to edit admin panel.
Description
The Cisco® SPA3102 Phone Adapter with Router features the ability to connect standard telephones and fax machines to an IP-based data network, with the additional benefit of an integrated connection for legacy telephone network "hop-on, hop-off" applications. SPA3102 users will be able to extend the use of their broadband phone service by automatically routing local calls from mobile phones and landlines to voice over IP (VoIP) service providers, and vice versa. If power is lost to the unit or Internet service is down, calls can be redirected to a traditional carrier via the FXO interface.

A user calling from a mobile phone or landline will be able to reduce and even eliminate international and long-distance telephone charges by first calling the Cisco SPA3102 via a local telephone number. The advanced authentication and call-routing intelligence programmed into the SPA3102 will route the call via the Internet to the end destination. In addition, when using a SPA3102 at the far end, VoIP calls placed to that location can be either answered or further processed and routed on as local calls to any legacy land line or mobile phone.

The Cisco SPA3102 supports one RJ-11 basic telephone FXS port to connect an existing analog phone or fax machine. It also supports one public switched telephone network (PSTN) FXO port to connect to a telephone company (Telco) or private branch exchange (PBX) circuit. The SPA3102 also includes two 100BASE-T RJ-45 Ethernet interfaces to connect to a home or office LAN, as well as an Ethernet connection to a broadband modem or router. The FXS and FXO lines can be configured independently via software controlled by the service provider or the end user.

Installed by the end user and remotely provisioned, configured, and maintained by the service provider, each Cisco SPA3102 converts voice traffic into data packets for transmission over an IP network. Compact in design, the SPA3102 can be used in consumer and business VoIP service offerings, including a full-featured IP Centrex environment. The SPA3102 uses international standards for voice and data networking for reliable voice and fax voice and carrier-grade feature support: The Cisco SPA3102 delivers clear, high-quality voice communication in diverse network conditions. Excellent voice quality in a demanding IP network is achieved via the advanced implementation of standard voice coding algorithms. The SPA3102 is interoperable with common telephony equipment such as voicemail, fax, PBX, and interactive voice response systems.
Large-scale deployment and management: The Cisco SPA3102 enables service providers to provide customized VoIP services to their subscribers. It can be remotely provisioned and supports dynamic, in-service software upgrades. A highly secure profile upload saves providers the time and expense of managing and preconfiguring or reconfiguring customer premises equipment (CPE).
Ironclad security: Cisco understands that security for end users and service providers is a fundamental requirement for a solid, carrier-grade telephony service. The Cisco SPA3102 supports highly secure, standard encryption-based methods for communication, provisioning, and servicing.Tech Specs

Data networking:
MAC address (IEEE 802.3)
IPv4 (RFC 791) upgradeable to v6 (RFC 1883)
Address Resolution Protocol (ARP)
DNS A record (RFC 1706), SRV record (RFC 2782)
Dynamic Host Configuration Protocol (DHCP) client (RFC 2131)
DHCP server (RFC 2131)
Point-to-Point Protocol over Ethernet (PPoE) client (RFC 2516)
Internet Control Message Protocol (ICMP) (RFC 792)
TCP (RFC 793)
User Datagram Protocol (UDP( (RFC 768)
Real Time Protocol (RTP) (RFC 1889, 1890)
Real Time Control Protocol (RTCP) (RFC 1889)
Differentiated Services (DiffServ) (RFC 2475), type of service (ToS) (RFC 791, 1349)
VLAN tagging 802.1p
Simple Network Time Protocol (SNTP) (RFC 2030)
Upload data rate limiting - static and automatic
Quality of service (QoS) - voice packet prioritization over other packet types
Router or bridge mode of operation
MAC address cloning
Port forwarding

Voice gateway:
Session Initiation Protocol (SIP) v2 (RFC 3261, 3262, 3263, 3264)
SIP proxy redundancy - dynamic via DNS SRV, A records
Reregistration with primary SIP proxy server
SIP support in Network Address Translation (NAT) networks (including Serial Tunnel [STUN])
Secure (encrypted) calling via prestandard implementation of Secure RTP
Codec name assignment

Voice algorithms:
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 capability:
Fax tone detection and pass-through (using G.711)
Fax pass-though (using G.711)
Dual-tone multifrequency (DTMF): in-band and 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)
Voice activity detection (VAD) with silence suppression
Attenuation/gain adjustments
Flash hook timer
Message waiting indicator (MWI) tones
Visual MWI (VMWI) via frequency shift keying (FSK)
Polarity control
Hook flash event signaling
Caller ID generation (name and number) - Bellcore, DTMF, European Telecommunications Standards Institute (ETSI)
Music on hold client
Streaming audio server - up to 10 sessions

Security:
Password-protected system reset to factory default
Password-protected administrator and user access authority
HTTPS with factory-installed client certificate
HTTP digest - encrypted authentication via MD5 (RFC 1321)
Up to 256-bit Advanced Encryption Standard (AES) encryption

Provisioning, administration, and maintenance:
Web browser administration and configuration via integrated web server
Telephone keypad configuration with interactive voice prompts
Automated provisioning and upgrade via HTTP, Trivial File Transfer Protocol (TFTP)
Asynchronous notification of upgrade availability via SIP NOTIFY
Nonintrusive, in-service upgrades
Report generation and event logging
Stats in BYE message
Syslog and debug server records - per-line configurable
Per line and purpose configurable syslog and debug options

Physical interfaces:
Two 100BASE-T RJ-45 Ethernet ports (IEEE 802.3) - 1 WAN, 1 LAN
1 RJ-11 FXS phone port - for analog circuit telephone device (tip/ring)
1 RJ-11 FXO phone port - for a Telco or PBX connection

FXS: Ring voltage: 40-55 VRMs configurable

Subscriber line interface circuit (SLIC):
Ring frequency: 10-40 Hz
Ring waveform: trapezoidal and sinusoidal
Maximum ringer load: 3 ringer equivalence numbers (RENs)
On-hook/off-hook characteristics:
On-hook voltage (tip/ring): -50V nominal
Off-hook current: 25 mA min
Terminating impedance: 8 configurable settings including North America 600 ohms, European CTR21

Regulatory compliance: FCC (Part 15 Class B), CE, ICES-003, A-Tick certification, RoHS

Power supply:
DC input voltage: +5V DC at 2.0A max
Power consump

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