Administrator’s Guide SoundPoint IP / SoundStation IP

This configuration attribute is defined as follows:

Attribute

Permitted Values

Default

Interpretation

 

 

 

 

dialplan.digitmap

string compatible with the

[2-9]110T

When this attribute is

 

digit map feature of

+011xxx.T

present, number-only dialing

 

MGCP described in 2.1.5

during the setup phase of

 

0[2-9]xxxxxxxxx

 

of RFC 3435. String is

new calls will be compared

 

+1[2-9]xxxxxxxx

 

limited to 768 bytes and

against the patterns therein

 

30 segments; a comma is

[2-9]xxxxxxxxx

and if a match is found, the

 

also allowed; when

[2-9]xxxT

call will be initiated

 

reached in the digit map,

automatically eliminating the

 

 

 

a comma will turn dial

 

need to press Send.

 

tone back on;’+’ is allowed

 

Attributes

 

as a valid digit; extension

 

 

 

dialplan.applyToCallLis

 

letter ‘R’ is used as

 

 

 

tDial,

 

defined above.

 

 

 

dialplan.applyToDirecto

 

 

 

ryDial,

 

 

 

dialplan.applyToUserDia

 

 

 

l, and

 

 

 

dialplan.applyToUserSen

 

 

 

d control the use of match

 

 

 

and replace in the dialed

 

 

 

number in the different

 

 

 

scenarios.

 

 

 

 

dialplan.digitmap.timeOut

string of positive integers

3 3 3 3 3 3

Timeout in seconds for each

 

separated by ‘’

 

segment of digit map.

 

 

 

Note: If there are more digit

 

 

 

maps than timeout values,

 

 

 

the default value of 3 will be

 

 

 

used. If there are more

 

 

 

timeout values than digit

 

 

 

maps, the extra timeout

 

 

 

values are ignored.

 

 

 

 

Routing <routing/>

This attribute allows the user to create a specific routing path for outgoing SIP calls independent of other “default” configurations.

This attribute also includes:

Server <server/>

Emergency <emergency/>

A - 20

Page 174
Image 174
Polycom SIP 3.1 manual Server server Emergency emergency, Attribute Permitted Values Default Interpretation

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.