Cisco Small Business Analog Adapter SPA112 with 2 FXS Ports, 77,47 €

The Cisco SPA112 2 Port Adapter enables high-quality VoIP service with a comprehensive feature set through a broadband Internet connection. Easy to install and use, it works over an IP network to connect analog phones and fax machines to a VoIP service provider and provides support for additional LAN connections.

The Cisco SPA112 includes two standard telephone ports to connect existing analog phones or fax machines to a VoIP service provider. Each phone line can be configured independently. With the Cisco SPA112, users can protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines as well as control their migration to IP voice with an extremely affordable, reliable solution.

Compact in design and compatible with international voice and data standards, the Cisco SPA112 can be used with residential, home-office, and small business VoIP service offerings, including full-featured hosted or open source IP PBX environments. This easy-to-use solution delivers advanced features to better connect employees and serve customers, all on a highly secure Cisco network.

The Cisco SPA112 2 Port Adapter:

Enables high-quality VoIP service with a comprehensive feature set through a broadband Internet connection
Provides high-quality, clear-sounding voice, using advanced voice quality-of-service (QoS) capabilities and the industry-leading voice Session Initiation Protocol (SIP) stack
Supports reliable faxing with simultaneous voice and data use
Includes two standard telephone ports, each with an independent phone number, for use with fax machines or analog phone devices
Is compatible with all industry voice and data standards and common telephone features such as caller ID, call waiting, and voicemail
Includes a simple-to-use web-based configuration utility for easy deployment

The Cisco SPA112 provides an easy-to-use VoIP solution that offers:

Toll-quality voice and carrier-class feature support: The Cisco SPA112 delivers clear, high-quality voice communication under a variety of network conditions. Excellent voice quality in challenging, changeable IPnetwork environments is made possible through the advanced implementation of standard voice codingalgorithms. The Cisco SPA112 is interoperable with common telephony equipment such as fax, voicemail,private branch exchanges (PBXs) and key telephone systems (KTSs), and interactive voice responsesystems.
Large-scale deployment and management: The Cisco SPA112 enables service providers to provide customized services to their subscribers. It can be remotely provisioned and supports dynamic, in-servicesoftware upgrades. A highly secure profile upload saves providers the time and expense of managing andpreconfiguring or reconfiguring customer premises equipment (CPE) for deployment.
Outstanding security: The Cisco SPA112 supports highly secure, encryption-based methods for communication, provisioning, and servicing.
Compact size: Designed for small spaces, the Cisco SPA112 can be installed as a desktop unit or mounted on a wall.
Comprehensive feature set: The standards-based Cisco SPA112 is compatible with Internet VoIP provider features such as caller ID, call waiting, voicemail, call forwarding, distinctive ring, and much moreto provide a complete, affordable, and highly reliable solution for high-quality VoIP.
Easy installation and changes: The web-based configuration utility enables quick deployment and easy changes.
Investment protection: Businesses that are growing rapidly can use the solution with other Cisco Unified Communications solutions, providing industry-leading investment protection.
Peace of mind: Cisco solutions deliver the solid reliability you expect from Cisco. All solution components have been rigorously tested to help ensure easy setup, interoperability, and performance.

Data networking

MAC address (IEEE 802.3)
IPv4 (RFC 791) upgradeable to IPv6 (RFC 1883)
Address Resolution Protocol (ARP)
Domain Name System (DNS) A record (RFC 1706) and SRV record (RFC 2782)
Dynamic Host Configuration Protocol (DHCP) client (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) (RFC 1890)
Real Time Control Protocol (RTCP) (RFC 1889)
VLAN tagging (IEEE 802.1p)
Simple Network Time Protocol (SNTP) (RFC 2030)
SIP channels support for both UDP and TCP transport

Voice gateway

SIPv2 (RFC 3261, 3262, 3263, and 3264)
SIP proxy redundancy: Dynamic through use of DNS SRV A records
Reregistration with primary SIP proxy server
SIP support in network address translation (NAT) networks (including Serial Tunnel [STUN])
Highly secure (encrypted) calling using Secure RTP (SRTP)
Codec name assignment
G.711 (A-law and μ-law)
G.726 (32 kbps)
G.729 A
Dynamic payload
Adjustable audio frames per packet
Dual-tone multifrequency (DTMF): In-band and out-of-band (RFC 2833) (SIP information)

Voice features

Independent configurable dial plans with interdigit timers and IP dialing (1 per port)
Call progress tone generation
Jitter buffer: Adaptive
Frame loss concealment
Full-duplex audio
Echo cancellation (G.165 and G.168)
Voice activity detection (VAD)
Silence suppression
Comfort noise generation (CNG)
Attenuation and gain adjustments
Flash hook timer
Message waiting indicator (MWI) tones
Visual messaging waiting indicator (VMWI) using frequency shift keying (FSK)
Polarity control
Hook flash event signaling
Caller ID generation (name and number): Bellcore, DTMF, and European Telecommunications Standards
Institute (ETSI)
Streaming audio server: Up to 10 sessions
Music on hold
Call waiting, call waiting and caller ID
Caller ID with name and number
Caller ID blocking
Selective and anonymous call rejection
Call forwarding: No answer, busy, and all
Do not disturb
Call transfer, call return, and call back on busy
Three-way conference calling with local mixing
Per-call authentication and associated routing
Call blocking with toll restriction
Distinctive ringing: Calling and called number
Off-hook warning tone
Advanced inbound and outbound call routing
Hotline and warmline calling
Long silence (configurable time setting) silence threshold
Disconnect tone (for example, reorder tone)
Configurable ring frequency
Ring validation time setting
Tip and ring voltage adjustment setting
Ring indication delay setting

Fax capability

Fax tone detection pass-through
Fax pass-through using G.711
Real-time fax over IP using T.38 fax relay (T.38 support is dependent on fax machine and network and transport
resilience)

Security

Password-protected system reset to factory default
Password-protected administrator and user access authority
Provisioning, configuration, and authentication
HTTPS with factory-installed client certificate
HTTP digest: Encrypted authentication using MD5 (RFC 1321)
Up to 256-bit Advanced Encryption Standard (AES) encryption
SIP Transport Layer Security (TLS)

Provisioning, administration, and maintenance

Web browser administration and configuration using integral web server
Telephone keypad configuration with interactive voice prompts
Automated provisioning and upgrade using HTTPS, HTTP, and Trivial File Transfer Protocol (TFTP)
TR-069
Asynchronous notification of upgrade availability using Notify
Nonintrusive, in-service upgrades
Report generation and event logging
Statistics in Bye message
Debug server records and syslog: Per-line configurable web browser
Ping and traceroute diagnostics
Configuration management: Backup and restore
Support for Bonjour

Physical interfaces