Configuration Files

 

 

 

 

 

 

Busy <busy/>

 

 

 

Calls can be automatically diverted when the phone is busy.

 

 

 

 

 

Attribute

 

Permitted Values

Default

Interpretation

 

 

 

 

 

divert.busy.x.enabled

 

0, 1

1

If set to 1, calls will be

 

 

 

 

forwarded on busy to the

 

 

 

 

contact specified below.

 

 

 

 

Note: If server-based call

 

 

 

 

forwarding is enabled, this

 

 

 

 

parameter is disabled.

 

 

 

 

 

divert.busy.x.timeout

 

positive integer

60

Time in seconds to allow

 

 

 

 

altering before initiating the

 

 

 

 

diversion.

 

 

 

 

 

divert.busy.x.contact

 

ASCII encoded string

Null

Forward-to contact for calls

 

 

containing digits (the user part

 

forwarded due to busy status, if

 

 

of a SIP URL) or a string that

 

Null, divert.x.contact will be

 

 

constitutes a valid SIP URL

 

used.

 

 

(6416 or 6416@polycom.com

 

 

 

 

 

 

 

 

No Answer <noanswer/>

 

 

 

The phone can automatically divert calls after a period of ringing.

 

 

 

 

 

Attribute

 

Permitted Values

Default

Interpretation

 

 

 

 

 

divert.noanswer.x.enabled

 

0, 1

1

If set to 1, calls will be

 

 

 

 

forwarded on no answer to the

 

 

 

 

contact specified.

 

 

 

 

Note: If server-based call

 

 

 

 

forwarding is enabled, this

 

 

 

 

parameter is disabled.

 

 

 

 

 

divert.noanswer.x.timeout

 

positive integer

60

Time in seconds to allow

 

 

 

 

altering before initiating the

 

 

 

 

diversion.

 

 

 

 

 

divert.noanswer.x.contact

 

ASCII encoded string

Null

Forward-to contact used for

 

 

containing digits (the user part

 

calls forwarded due to no

 

 

of a SIP URL) or a string that

 

answer, if Null,

 

 

constitutes a valid SIP URL

 

divert.x.contact will be

 

 

(6416 or 6416@polycom.com)

 

used.

 

 

 

 

 

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Polycom SIP 3.1 manual Calls can be automatically diverted when the phone is busy, Divert.x.contact will be

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.

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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.