Administrator’s Guide SoundPoint IP / SoundStation IP

This configuration attribute is defined as follows:

 

Permitted

 

 

Attribute

Values

Default

Interpretation

 

 

 

 

dialplan.x.digitmap

A string compatible with the

Null

When present, this attribute

 

digit map feature of MGCP

 

overrides the global dial plan

 

described in 2.1.5 of RFC

 

defined in the sip.cfg

 

3435; string is limited to 768

 

configuration file.

 

bytes and 30 segments; a

 

For more information, refer to

 

comma is also allowed; a

 

 

 

Digit Map <digitmap/> on page

 

comma is also allowed;

 

 

 

A-18.

 

when reached in the digit

 

 

 

 

 

map, a comma will turn dial

 

 

 

tone back on;’+’ is allowed

 

 

 

as a valid digit; extension

 

 

 

letter ‘R’ is used as defined

 

 

 

above.

 

 

 

 

 

 

dialplan.x.digitmap.timeOut

string of positive integers

Null

When present, and if

 

separated by ‘’

 

dialplan.x.digitmap is not

 

 

 

Null, this attribute overrides the

 

 

 

global dial plan defined in the

 

 

 

sip.cfg configuration file.

 

 

 

For more information, refer to

 

 

 

Digit Map <digitmap/> on page

 

 

 

A-18.

 

 

 

 

Routing <routing/>

This attribute allows specific routing paths for outgoing SIP calls to be configured independent of other ‘default’ configuration.

This attribute includes:

Server <server/>

Emergency <emergency/>

Server <server/>

This configuration attribute is defined as follows:

Attribute

Permitted Values

Default

Interpretation

 

 

 

 

dialplan.x.routing.server.y.

dotted-decimal IP address

Null

IP address or host name and

address

or host name

 

port of a SIP server that will

 

 

 

be used for routing calls.

dialplan.x.routing.server.y.port

1 to 65535

5060

Multiple servers can be listed

 

 

 

starting with y=1, 2, ... for

 

 

 

fault tolerance.

 

 

 

 

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Polycom SIP 3.1 manual This configuration attribute is defined as follows, Digit Map digitmap/ on

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.