Configuring Your System

Distinctive Ringing

There are three options for distinctive ringing:

1.The user can select the ring type for each line. This option has the lowest priority.

2.The ring type for specific callers can be assigned in the contact directory. For more information, refer to Distinctive Incoming Call Treatment, the previous section. This option has a higher priority than option 1 and a lower priority than option 3.

3.The voIpProt.SIP.alertInfo.x.value and voIpProt.SIP.alertInfo.x.class fields can be used to map calls to specific ring types. This option has the highest priority.

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

Central

Configuration file:

Specify the mapping of Alert-Info strings to ring types.

(boot server)

sip.cfg

For more information, refer to Alert Information <alertInfo/> on

 

 

 

page A-15.

 

 

 

 

Configuration file:

Specify the ring type to be used for each line.

 

phone1.cfg

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

 

 

 

 

XML File: <Ethernet

This file can be created manually using an XML editor.

 

address>-directory.

For more information, refer to Local Contact Directory on page

 

xml

 

4-9.

 

 

 

Local

Local Phone User

The user can edit the ring types selected for each line under the

 

Interface

Settings menu. The user can also edit the directory contents.

 

 

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.

 

 

 

 

Distinctive Call Waiting

The voIpProt.SIP.alertInfo.x.value and voIpProt.SIP.alertInfo.x.class fields can be used to map calls to distinct call waiting types, currently limited to two styles.

Configuration changes can performed centrally at the boot server:

Central (boot server)

Configuration file: sip.cfg

Specify the mapping of Alert-Info strings to call waiting types.

For more information, refer to Alert Information <alertInfo/> on page A-15.

4 - 7

Page 61
Image 61
Polycom SIP 3.1 manual Distinctive Ringing, Distinctive Call Waiting, Address-directory, Xml Local

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.