Administrator’s Guide SoundPoint IP / SoundStation IP

The individual parties within a conference can be managed. New parties can be added and information about the conference participants can be viewed (for example, names, phone numbers, send/receive status or media flow, receive and transmit codecs, and hold status).

Configuration changes can be performed centrally at the boot server:

Central

Configuration file:

Turn this feature on or off.

(boot server)

sip.cfg

• For more information, refer to Feature <feature/> on page A-92.

 

 

 

Call Forward

The phone provides a flexible call forwarding feature to forward calls to another destination. Call forwarding can be applied in the following cases:

•Automatically to all calls

•Calls from a specific caller (extension)

•When the phone is busy

•When Do Not Disturb is active

•After an extended period of alerting

The user can elect to manually forward calls while they are in the alerting state to a predefined or manually specified destination. The call forwarding feature works in conjunction with the distinctive incoming call treatment feature (refer to Distinctive Incoming Call Treatment on page 4-6). The user’s ability to originate calls is unaffected by all call forwarding options. Each registration has its own forwarding properties.

Server-based call forwarding is active if the feature is enabled on both the phone and the server and the phone is registered. If server-based call forwarding is enabled on any of the phone’s registrations, the other registrations are not affected.

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

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

•If server-based call forwarding is enabled, but inactive, and the user selects the call forward soft key, the “moving arrow” icon does not appear on the user’s phone (incoming calls are not forwarded).

Note

Server-based call forwarding is disabled if Shared Call Appearance or Bridged Line

 

Appearance is enabled.

 

 

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Polycom SIP 3.1 manual Call Forward
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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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