Administrator’s Guide SoundPoint IP / SoundStation IP

This attribute also includes:

Configuration <cfg/>

Configuration <cfg/>

This configuration attribute is defined as follows:

Attribute

Permitted

Default

Interpretation

Values

 

 

 

 

httpd.cfg.enabled

0, 1

1

If set to 1, the HTTP server configuration interface

 

 

 

will be enabled.

 

 

 

 

httpd.cfg.port

1-65535

80

Port is 80 for HTTP servers. Care should be taken

 

 

 

when choosing an alternate port.

 

 

 

 

Call Handling Configuration <call/>

This configuration attribute is defined as follows:

 

Permitted

 

 

Attribute

Values

Default

Interpretation

 

 

 

 

call.rejectBusyOnDnd

0, 1

1

If set to 1, reject all incoming calls with the

 

 

 

reason “busy” if do-not-disturb is enabled.

 

 

 

Note: This attribute is ignored when the line is

 

 

 

configured as shared. The reason being that

 

 

 

even though one party has turned on DND, the

 

 

 

other person/people sharing that line do not

 

 

 

necessarily want all calls to that number diverted

 

 

 

away.

 

 

 

Note: If server-based DND is enabled, this

 

 

 

parameter is disabled.

 

 

 

 

call.enableOnNotRegistered

0, 1

1

If set to 1, calls will be allowed when the phone is

 

 

 

not successfully registered, otherwise, calls will

 

 

 

not be permitted without a valid registration.

 

 

 

 

call.offeringTimeOut

positive

60

Time in seconds to allow an incoming call to ring

 

integer

 

before dropping the call, 0=infinite.

 

 

 

Note: The call diversion, no answer feature will

 

 

 

take precedence over this feature if enabled. For

 

 

 

more information, refer to No Answer

 

 

 

<noanswer/> on page A-115.

 

 

 

 

call.ringBackTimeOut

positive

60

Time in seconds to allow an outgoing call to

 

integer

 

remain in the ringback state before dropping the

 

 

 

call, 0=infinite.

 

 

 

 

A - 64

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Polycom SIP 3.1 manual Call Handling Configuration call, Configuration cfg
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SIP 3.1 specifications

Polycom SIP 3.1 is an advanced session initiation protocol designed to enhance voice and video communication in various business environments. As a pivotal component of Polycom’s telecommunication solutions, SIP 3.1 offers several features and characteristics that cater to the evolving needs of modern enterprises, particularly those that rely on seamless and efficient communication.

One of the standout features of Polycom SIP 3.1 is its robust interoperability. This protocol supports a wide range of endpoints and platforms, allowing organizations to integrate their existing systems with new technologies effortlessly. This flexibility ensures that businesses can leverage their previous investments while upgrading to the latest communication tools.

Another key aspect of Polycom SIP 3.1 is its enhanced call management capabilities. The protocol facilitates efficient call handling, enabling users to manage multiple calls seamlessly. Features like call hold, transfer, and conferencing are streamlined, which enhances productivity by allowing for efficient collaboration among team members. Additionally, it is optimized for low latency and high-quality audio, ensuring that conversations are clear and free from disruptions.

Security is paramount in today’s digital landscape, and Polycom SIP 3.1 addresses this concern with advanced encryption standards. By integrating security features such as Transport Layer Security (TLS) and Secure Real-time Transport Protocol (SRTP), it protects sensitive communication from unauthorized access and ensures that data remains confidential throughout the call.

Polycom SIP 3.1 also boasts compatibility with various video codecs, making it a versatile choice for video conferencing. This compatibility ensures high-quality video streams, which is essential for effective visual communication in remote meetings. Furthermore, the support for the H.264 codec provides efficient bandwidth usage, making high-definition video conferencing accessible, even in varying network conditions.

Moreover, the protocol provides strong support for presence and instant messaging, which enhances real-time communication among users. This integration of voice, video, and messaging capabilities fosters a more connected and collaborative work environment, allowing teams to engage effectively regardless of their geographical locations.

In summary, Polycom SIP 3.1 stands out as a sophisticated solution tailored to meet the demands of modern business communication. With its emphasis on interoperability, call management, security, video quality, and real-time collaboration, it caters to companies of all sizes seeking to optimize their communication infrastructure and enhance productivity in the workplace. As businesses continue to navigate the complexities of digital communication, Polycom SIP 3.1 remains a compelling choice in the market.
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