This is a Cisco/Linksys SPA192 Analog Telephone Adapter with Router that’s often used on our iDirect Satellite Internet Platform for VoIP Phone Service.
This model is a replacement for the SPA122Â that is no longer available!
The Cisco® ATA 192 Multiplatform Analog Telephone Adapter is a 2-port handset-to-Ethernet adapter that brings traditional analog devices into the IP world.
Product overview
The Cisco ATA 192 Multiplatform Analog Telephone Adapter turns traditional telephone, fax, and overhead paging communications devices into IP devices for greater cost-effectiveness. Customers can take advantage of IP telephony applications by connecting their analog devices to Cisco analog telephone adapters.
The ATA 192 is the preferred solution to address the needs of customers who connect to enterprise networks, small offices, or unified communications as a service from the cloud. It has two standard FXS ports, which can be configured independently as two Session Initiation Protocol (SIP) registrations. It also has two 100BASE-T ports with an integrated high-performance router to extend local network connectivity. With the ATA 192, customers can protect and extend their existing investment in analog systems, as well as smooth their migration to pure voice-over IP in a more affordable and reliable way.
Features and benefits
Feature | Benefit |
Voice quality | Offers clear, natural-sounding voice quality via advanced preprocessing, high-performance echo cancellation, voice activity detection, and comfort noise generation |
Cloud provisioning | Enables zero-touch provisioning via TR-069 and XML configuration files |
Security | Provides a complete security solution for both media and signaling |
Problem reporting (PRT) | Improves serviceability with a dedicated PRT button for problem reporting and log collection |
IPv6 | Enables IPv6 dual-stack to help with migration to IPv6 |
Feature | Specifications |
Physical dimensions (H×W×D) | 3.9 x 3.9 x 1.1 in. (100 × 100 × 28 mm) |
Weight (g) | 4.7 oz (132.1 g) |
Hardware | Interface: Two RJ-11 FXS ports, two 10/100 Mbps RJ-45 Ethernet ports Button: Reset / Problem Reporting (PRT) LED indicators: Power, Network, Phone 1, Phone 2, PRT Wall mountable |
Subscriber Line Interface Circuit (SLIC) | Ring voltage: 40 to 90 Vpk configurable Ring frequency accuracy: 1% Ring waveform: Trapezoidal or sinusoidal Maximum ringer load: 3 Ringer Equivalence Numbers (RENs) On-hook voltage (tip and ring): -46 to -56V Off-hook current: 25mA +/- 10% Terminating impedance: 600 ohms resistive, 900 ohms resistive, or 220 ohms + 820 ohms 120 nF complex impedance Frequency response: 300 to 3400 Hz Return loss (600 ohms, 300 to 3400 Hz): up to 26 dB Idle channel noise: <-65 dBm 0p Longitudinal balance: 58 dB (typical) Voice quality Mean Opinion Score (MOS): >4.0 Voice quality jitter: <150 ms |
Networking | MAC address IPv4 only IPv6 only IPv4/IPv6 dual-stack Session Initiation Protocol (SIP) Transmission Control Protocol (TCP) User Datagram Protocol (UDP) Real-Time Protocol (RTP) Real-Time Control Protocol (RTCP) HTTP HTTPS Trivial File Transfer Protocol (TFTP) Address Resolution Protocol (ARP) DNS A/AAAA and SRV records Dynamic Host Configuration Protocol (DHCP) client Internet Control Message Protocol (ICMP) Simple Network Time Protocol (SNTP) Cisco Discovery Protocol Link Layer Discovery Protocol (LLDP) Point-to-Point Protocol over Ethernet (PPPoE) |
Routing | Routing and bridging Static and dynamic address assignment Network Address Translation (NAT) DHCP client reservation MAC address cloning Port forwarding DMZ mode VPN pass-through: IP Security (IPsec) Encapsulating Security Payload (ESP), Point-to-Point Tunneling Protocol (PPTP), Layer 2 Tunneling Protocol (L2TP) |
Quality of Service (QoS) | IEEE 802.1p/Q (QoS and VLAN tagging) Differentiated Services (DiffServ), Type of Service (ToS) |
Telephony | Anonymous call and call blocking Call forwarding: No answer, busy, and all Call hold and resume Caller ID blocking Caller ID generation (name and number): Bellcore, BT, and European Telecommunications Standards Institute (ETSI) Caller ID with name and number Call pickup and group pickup Call transfer, call return and call back on busy Call waiting Configurable ring frequency Configurable tones and cadences Disconnect tone Distinctive ringing: Calling and called number Do not disturb Forced Authorization Code (FAC)/Client Matter Code (CMC) Flash hook timer Hook flash event signaling Hotline and warm line calling Message Waiting Indicator (MWI) tones Music on hold Off-hook warning tone Polarity control Redial Selective and anonymous call rejection SIP redundancy Speed dial Streaming audio server: Up to 4 sessions Three-way conference calling with local mixing Tip and ring voltage adjustment setting Visual Messaging Waiting Indicator (VMWI) using frequency-shift keying (FSK) Network Address Translation (NAT) Session Traversal Utilities for NAT (STUN) |
Audio | Codec: G.711 a-law, G.711 μ-law, G.729a, G.729ab, G.726 Codec name assignment Full-duplex audio Echo cancellation Voice activity detection Silence suppression Configurable silence threshold Comfort noise generation Adaptive jitter buffer Frame loss concealment Adjustable audio frames per packet Call progress tone generation Impedance and gain adjustment Dynamic audio payload |
Fax | Real-time fax over IP via T.38 fax relay (Group 3) Fax pass-through via G.711 (Group 3) Fax tone detection and pass-through Automatic negotiation on the transmission rate |
Provisioning and management | Cloud provisioning (remote configuration) Web-based administration Interactive Voice Response (IVR) Automated provisioning and upgrading via HTTP, HTTPS, and TFTP TR-069 SSH access Simple Network Management Protocol (SNMPv3) Report generation and event logging Dedicated PRT button Support for RTP statistics Syslog (multilevel granularity) Ping and traceroute diagnostics Configuration management: Backup and restore Dual image |
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 Advanced Encryption Standard (AES) encryption SIP over Transport Layer Security (TLS1.1 and TLS1.2) Secure (encrypted) calling using Secure RTP (sRTP) Encrypted configuration files Image authentication Secure boot Secure Shell (SSH) |
Power | DC input voltage: 5V DC at 2.4A maximum Power consumption: 5W Switching type (100-240V) automatic Power adapter: 100-240V and 50-60 Hz (26-34 VA) AC input, with 1.8m cord |
Reliability | Mean Time Between Failures (MTBF): 300,000 hours Operating temperature: 32° to 104°F (0° to 40°C) Nonoperating temperature: 14° to 140°F (-10° to 60°C) Humidity: Operating 10% to 90%, noncondensing / nonoperating 10% to 95%, noncondensing |
Compliance (regulatory) | CE Markings per directives 2014/30/EU and 2014/35/EU |
Compliance (safety) | UL 60950 Second Edition CAN/CSA-C22.2 No. 60950 Second Edition IEC 60950-1:2005 (Second Edition) + A1:2009 + A2:2013 and/or AS/NZS 60950.1:2015 |
Compliance (EMC) | AS/NZS CISPR 32:2015 Class B CISPR 32: 2015 Class B EN 55032: 2015 Class B EN 61000-3-2: 2014 Class A EN 61000-3-3: 2013 EN 55024:2010+A1: 2015 EN 61000-4-2: 2009 EN 61000-4-3: 2006+A1:2008+A2:2010 EN 61000-4-4: 2012 EN 61000-4-5: 2014 EN 61000-4-6: 2014+AC：2015 EN 61000-4-8: 2010 EN 61000-4-11: 2004 FCC Part 15, Subpart B ANSI C63.4-2014 ICES-003 Issue 6: 2016 ANSI C63.4-2014 VCCI-TECHNICAL REQUIREMENTS (VCCI-CISPR 32: 2016) / CISPR 32: 2015 class B |
Feature | Benefit |
Voice quality | Offers clear, natural-sounding voice quality via advanced preprocessing, high-performance echo cancellation, voice activity detection, and comfort noise generation |
Cloud provisioning | Enables zero-touch provisioning via TR-069 and XML configuration files |
Security | Provides a complete security solution for both media and signaling |
Problem reporting (PRT) | Improves serviceability with a dedicated PRT button for problem reporting and log collection |
IPv6 | Enables IPv6 dual-stack to help with migration to IPv6 |
