Good Side of This Phone Adapter
The Cisco ATA190 is a VoIP (Voice over Internet Protocol) phone adapter that allows users to connect traditional analog phones to a VoIP network. It has two ports, one for the analog phone and one for the connection to the VoIP network. This Product is designed to work with a variety of analog phones, including single-line and multi-line phones, fax machines, and modems. It supports a range of features to help improve the quality of VoIP calls, including echo cancellation, jitter buffer, and automatic gain control. This Model is also equipped with security features such as encryption and authentication to help protect against unauthorized access.
The Cisco ATA190 2-Port VoIP Phone Adapter is designed to be compatible with a wide range of devices and networking equipment. It is compatible with a variety of analog phones, including single-line and multi-line phones, fax machines, and modems. This Product is also compatible with a range of VoIP network infrastructure equipment, including VoIP gateways, PBX systems, and SIP servers. It is compatible with a number of industry-standard protocols, including SIP, H.323, and MGCP, making it easy to integrate into existing VoIP networks. In terms of operating system compatibility, This Product is designed to work with a variety of platforms, including Windows, Linux, and Mac OS. It is also compatible with a number of Cisco networking software and tools, including Cisco Configuration Professional and Cisco Unified Communications Manager.
Quality Control Measures
The Cisco ATA190 UC ATA 190 2 Ports VoIP Phone Adapter is a device that allows analog phones to connect to a digital VoIP network. Quality control measures for this device are important to ensure that it functions correctly and delivers high-quality voice communication.
Here are some quality control measures for the Cisco ATA190 UC ATA 190 2 Ports VoIP Phone Adapter:
- Testing: The device should be tested thoroughly to ensure that it is working as intended. This includes testing the device with different types of analog phones and in different network configurations.
- Configuration: The device should be configured correctly to ensure that it is optimized for the specific network environment. This includes configuring network settings, voice codecs, and other settings that affect voice quality.
- Firmware: The device firmware should be updated to the latest version to ensure that any known issues or bugs are resolved.
- Quality of Service (QoS): QoS settings should be configured correctly to ensure that voice traffic is prioritized over other types of network traffic. This helps to minimize latency, jitter, and packet loss, which can all affect voice quality.
- Security: The device should be configured with appropriate security settings to prevent unauthorized access and ensure data privacy.
- Monitoring: The device should be monitored regularly to detect any issues that may arise. This includes monitoring network traffic, device logs, and other metrics that can indicate potential issues.
General Information about this Cisco ATA190
- Manufacturer: Cisco
- Part Number or SKU# ATA190
- Product Type: VoIP Gateway
Technical Information of VoIP Phone Adapter
- Form Factor: External
- Connectivity Technology: Wired
Interfaces/Ports for 2 Ports VoIP Adapter
- Serial Port: No
- PoE (RJ-45) Port: No
- USB: No
- Management Port: No
- Number of Network (RJ-45) Ports: 1
- Number of FXS (RJ-11) Ports: 2
Network & Communication
- Ethernet Technology: Fast Ethernet
- Network / Transport Protocol: DHCP
- Remote Management Protocol: TFTP
- Wireless LAN: No
Management & Protocols
- Voice Codec G.729a, G.729ab, G.711a, G.711u
- VoIP Protocols: SIP, SRTP
- Telephony Interfaces: 2 phone (FXS)
- IP Telephony Features: Comfort noise generation (CNG), voice activity detection (VAD), DTMF generation/detection
- Power Device: External power adapter
- Voltage Required: AC 120/230 V / DC 5 V
- Power Consumption Operational: 5 Watt
- Min Operating Temperature: 32 °F
- Max Operating Temperature: 113 °F
- Humidity Range Operating: 0 - 90% (non-condensing)
Dimensions & Weight
- Form Factor Desktop
- Height: 1.1 Inch
- Width 4 Inch
- Depth: 4 Inch
- Weight (Approximate) 0.33 Lbs
- Environmentally Friendly Yes
- Environmental Certification RoHS
About Refurbished Products
The voice-over internet protocol gateway essentially acts as a bridge between legacy telephone systems and VoIP solutions by intelligently adapting and routing calls from one medium to another. VoIP is the technology used to make phone calls via the internet rather than through mobile devices or regular landlines. A high-speed internet connection is required to support IP telephony. VoIP gateways take analog voice signals and convert them into data packets sent just like digital data over a broadband connection.
The gateway serves to connect traditional phone systems such as ACDs and PBXs with VoIP services or vice versa (connecting VoIP solutions to legacy phone systems). The process of converting voice-over IP calls from one code to another is called transcoding. Depending on the VoIP setup, you may be in a position to keep the old phone number.
Normally, gateways will be used to connect to multiple VoIP endpoints such as services and devices to ensure PSTN resources are shared between multiple IP communication systems.
The addition of VoIP networking to legacy phone systems presents the opportunity to add more features and extend the life of legacy equipment. The use of VoIP solutions alongside legacy equipment in a hybrid setup will help in cutting costs and preventing disruption of business processes. The legacy phone system will, therefore, see the gateway as another networked PBX or as the phone service provider. It works as follows: voice signals from phone calls using the PBX are converted into VoIP data and transmitted over the internet using the service provider or VoIP peers. The incoming calls from VoIP are converted into the legacy protocol being used and sent to the PBX.
There are two types of VoIP gateways:
- Analog gateways – used to convert between traditional analog phones and phone lines and VoIP protocols.
- Digital gateways – used to convert between VoIP protocols and different types of digital phone services such as E1, T1, BRI, and PRI.
The biggest benefit of VoIP phones is the cheaper cost of calling compared to landlines especially when calling over a long distance. Some setups are completely free or are bundled together as a package with FTTP connections.
Key features of our VoIP gateways include:
- Compliance with multiple protocols including H.323, MGCP, SCCP, and SIP.
- Automatic provisioning via TFTP/HTTP
- Support for many voice codecs including G.711, G.723.1, G.726, and G.729A.
- Call routing and least-cost call routing capability
- Echo cancellation, Jitter buffering, VAD, and CNG for clarity even on low-strength connections.
- T.38 compliance to support faxing.
- Web-based administration/management
VoIP gateways are also used to build redundancy on a system such that the gateway communicates with the primary IP in normal PBX setups. The gateway will communicate with a backup system if there is a failure. The use of a gateway to connect a VoIP system to a traditional phone makes sense if SIP trunks are unavailable or the reliability of a PSTN is needed.
VoIP gateways can also be used for staged migrations by setting it up as a bridge within a legacy PBX setup, the PSTN setup, and the new IP PBX setup. The PSTN setup trunks will be connected to one interface while the second interface will be connected to the legacy PBX trunk port. A VoIP protocol integration is undertaken for the new IP PBX.
The VoIP gateway also offers flexibility by passing calls between two PBXs and becomes useful in allowing some departments within a company to keep using a legacy system as other departments migrate to the VoIP system.We offer a big line of new and refurbished VoIP gateways including wireless, external, 10-100 gateways, and other VoIP solutions from reputable manufacturers that will meet your needs without exceeding your budget.
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