Administrator’s Guide SoundPoint IP / SoundStation IP

Do Not Disturb

A Do Not Disturb (DND) feature is available to temporarily stop all incoming call alerting. Calls can optionally be treated as though the phone is busy while DND is enabled. DND can be configured as a per-registration feature.

Incoming calls received while DND is enabled are logged as missed. For more information on forwarding calls while DND is enabled, refer to Call Forward on page 4-20.

Server-based DND is active if the feature is enabled on both the phone and the server and the phone is registered. The server-based DND feature is applicable for all registrations on the phone (no per-registration mode) and it disables local Call Forward and DND features.

Server-based DND will behave the same as per-SIP 2.1 per-registration feature with the following exceptions:

There is no indication on the phone’s user interface whether or not server-based DND is active.

If server-based DND is enabled, but inactive, and the user presses the DND key or selects the DND option on the Feature menu, the “Do Not Disturb” message does not appear on the user’s phone (incoming call alerting will continue).

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

Central

Configuration file:

Enable or disable server-based DND.

(boot server)

sip.cfg

For more information, refer to SIP <SIP/> on page A-10

 

 

Specify whether or not DND results in incoming calls being given

 

 

busy treatment.

 

 

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

 

 

on page A-64.

 

 

 

 

Configuration file:

Enable or disable server-based DND as a per-registration feature.

 

phone1.cfg

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

 

 

Specify whether DND is treated as a per-registration feature or a

 

 

global feature on the phone.

 

 

For more information, refer to Do Not Disturb <dnd/> on page

 

 

A-116.

 

 

 

Local

Local Phone User

Enable or disable DND using the “Do Not Disturb” key on the

 

Interface

SoundPoint IP 301, 501, 550, 560, 600, 601, and 650 and 670 or the

 

 

“Do Not Disturb” option on the Features menu on the SoundPoint IP

 

 

320, 330, and 430 and SoundStation IP 4000, 6000, and 7000.

 

 

 

Handset, Headset, and Speakerphone

SoundPoint IP phones come standard with a handset and a dedicated connector is provided for a headset (not supplied). The SoundPoint IP 320, 330, 430, 500, 501, 550, 560, 600, 601, 650, and 670 desktop phones and SoundStation

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Polycom SIP 3.1 manual Do Not Disturb, Handset, Headset, and Speakerphone, 116

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.