Administrator’s Guide SoundPoint IP / SoundStation IP

simultaneously. This feature is dependent on support from a SIP server that binds the appearances together logically and looks after the necessary state notifications and performs an access control function. For more information, refer to Bridged Line Appearance Signaling on page B-10.

 

Note

In the configuration files, bridged lines are configured by “shared line” parameters.

 

 

 

 

 

 

 

 

Configuration changes can performed centrally at the boot server or locally:

 

 

 

 

Central

 

Configuration file:

Specify whether diversion should be disabled on shared lines.

(boot server)

 

sip.cfg

 

For more information, refer to Call Handling Configuration <call/>

 

 

 

 

on page A-64.

 

 

 

 

 

 

Configuration file:

Specify per-registration line type (private or shared) and the shared

 

 

phone1.cfg

line third party name. A shared line will subscribe to a server

 

 

 

 

providing call state information.

 

 

 

 

For more information, refer to Registration <reg/> on page A-107.

 

 

 

 

Specify per-registration whether diversion should be disabled on

 

 

 

 

shared lines.

 

 

 

 

For more information, refer to Diversion <divert/> on page A-114.

 

 

 

 

Local

 

Web Server

Specify per-registration line type (private or shared) and third party

 

 

(if enabled)

name, and whether diversion should be disabled on shared lines.

 

 

Navigate to http://<phoneIPAddress>/reg.htm

 

 

 

 

 

 

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

 

 

address>-phone.cfgon the boot server. Changes will permanently

 

 

 

 

override global settings unless deleted through the Reset Local

 

 

 

 

Config menu selection and the <Ethernet address>-phone.cfgis

 

 

 

 

removed from the boot server.

 

 

 

 

 

 

Local Phone User

Specify per-registration line type (private or shared) and the shared

 

 

Interface

 

line third party name using the SIP Configuration menu. Either the

 

 

 

 

Web Server or the boot server configuration files or the local phone

 

 

 

 

user interface should be used to configure registrations, not a mixture

 

 

 

 

of these options. When the SIP Configuration menu is used, it is

 

 

 

 

assumed that all registrations use the same server.

 

 

 

 

 

Busy Lamp Field

Note

This feature is available only on SoundPoint IP 320/330, 430, 550, 560, 600, 601,

 

650, and 670 phones. However, on the SoundPoint IP 320/330, the LED is not lit.

 

Depending on your call server, certain aspects of this feature work may not work as

 

described below.

 

 

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Polycom SIP 3.1 manual Busy Lamp Field

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.