ATA Voice Field Reference

SIP page

B

Debug Level

Determines the level of debug information that is generated. Select 0, 1, 2, or 3 from the drop-down menu. The higher the debug level, the more debug information is generated.

The default is 0, which indicates that no debug information is generated.

SIP page

You can use the Voice tab > SIP page to configure the SIP settings. With some variations, depending on the model, this page includes the following sections:

•”SIP Parameters section” section on page133

•”SIP Timer Values (sec) section” section on page135

•”Response Status Code Handling section” section on page137

•”RTP Parameters section” section on page138

•”SDP Payload Types section” section on page140

•”NAT Support Parameters section” section on page141

•”Trunking Parameters section (SPA8000)” section on page144

Voice tab > SIP page >

SIP Parameters section

Max Forward

SIP Max Forward value, which can range from 1 to 255.

 

The default is 70.

 

 

Max Redirection

Number of times an invite can be redirected to avoid an

 

infinite loop.

 

The default is 5.

 

 

Max Auth

Maximum number of times (from 0 to 255) a request may

 

be challenged.

 

The default is 2.

 

 

ATA Administration Guide

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Cisco Systems SPA2102, SPA3102, WRP400, SPA8000, PAP2T manual Sip, SIP Parameters section
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PAP2T, SPA8000, SPA3102, WRP400, SPA2102 specifications

The Cisco Systems SPA2102 is a versatile Voice over Internet Protocol (VoIP) adapter that serves as a bridge between traditional telephony systems and modern IP networks. Designed primarily for small to medium businesses, the SPA2102 is highly regarded for its reliability, ease of use, and rich feature set. This device allows users to make and receive phone calls over the internet while maintaining the ability to connect traditional analog phones.

One of the standout features of the SPA2102 is its dual-port capability. The device includes two FXS ports, allowing users to connect two separate analog telephones. This makes it an ideal choice for businesses that want to retain their existing telephony infrastructure while transitioning to a VoIP system. The ability to utilize two telephone lines simultaneously provides flexibility and convenience, especially for users in a busy office environment.

The SPA2102 leverages Session Initiation Protocol (SIP) technology, which is widely recognized for its robustness and interoperability. The support for SIP ensures that the SPA2102 can work seamlessly with various VoIP service providers, offering users a broad range of options for their telecommunication needs. In addition to SIP, the device supports various codecs, including G.711, G.726, and G.729, allowing for flexible audio quality settings and bandwidth management.

Security is also a critical aspect of the SPA2102. It incorporates advanced encryption methods, such as Secure Real-time Transport Protocol (SRTP) and Transport Layer Security (TLS), to protect voice communications from potential eavesdropping. This focus on security makes the SPA2102 a reliable choice for businesses concerned about the confidentiality of their conversations.

The device is easy to configure and manage, thanks to its web-based interface. This allows administrators to quickly set up the adapter, manage network settings, and troubleshoot any issues that may arise. Furthermore, the SPA2102 supports Quality of Service (QoS) features, ensuring that voice traffic is prioritized over other types of network traffic, which enhances call quality and reliability.

Overall, the Cisco SPA2102 is a powerful, user-friendly VoIP adapter that combines traditional telephony with modern IP technology. Its dual-port capability, support for SIP, extensive security features, and ease of configuration make it a solid choice for businesses looking to innovate their communication systems while minimizing disruption. Whether used in a small office or a larger organizational setting, the SPA2102 continues to be a reliable component of VoIP solutions.
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