Configuring Your System

Call Log

The phone maintains a call log. The log contains call information such as remote party identification, time and date, and call duration. It can be used to redial previous outgoing calls, return incoming calls, and save contact information from call log entries to the contact directory.

The call log is stored in volatile memory and is maintained automatically by the phone in three separate lists: Missed Calls, Received Calls and Placed Calls. The call lists can be cleared manually by the user and will be erased when the phone is restarted.

 

Note

On some SoundPoint IP platforms, missed calls and received calls appear in one

 

 

 

list. Missed calls appear as

 

and received calls appear as

 

.

 

 

 

 

 

 

 

 

 

 

 

 

The “call list” feature can be disabled on all SoundPoint IP and SoundStation IP

 

 

 

platforms except the SoundPoint IP 330/320 and SoundStation IP 7000.

 

 

 

 

 

 

 

 

 

 

Configuration changes can performed centrally at the boot server:

 

 

 

 

 

 

Central

 

Configuration File:

Enable or disable all call lists or individual call lists.

 

 

(boot server)

 

sip.cfg

 

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

 

 

 

 

 

 

 

 

 

Call Timer

A call timer is provided on the display. A separate call timer is maintained for each distinct call in progress. The call duration appears in hours, minutes, and seconds.

There are no related configuration changes.

Call Waiting

When an incoming call arrives while the user is active on another call, the incoming call is presented to the user visually on the LCD display. A configurable sound effect such as the familiar call-waiting beep will be mixed with the active call audio as well.

Configuration changes can performed centrally at the boot server:

Central

Configuration File:

Specify the ring tone heard on an incoming call when another call is

(boot server)

phone1.cfg

active.

 

 

• For more information, refer to Call Waiting <callWaiting/> on page

 

 

A-113.

 

 

 

For related configuration changes, refer to Customizable Audio Sound Effects on page 4-5.

4 - 3

Page 56 Page 58
Page 57
Image 57
Polycom SIP 3.1 manual Call Log, Call Timer, Call Waiting
Page 56 Page 58

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.
Top Page Image Contents