Administrator’s Guide SoundPoint IP / SoundStation IP

 

Permitted

 

 

Attribute

Values

Default

Interpretation

 

 

 

 

dialplan.applyToUserSend

0, 1

1

This attribute covers the case when the user

 

 

 

presses the Send soft key to send the dialed

 

 

 

number.

 

 

 

Value interpretation is the same as for

 

 

 

dialplan.applyToCallListDial.

dialplan.impossibleMatchHandling

0, 1 or 2

0

If set to 0, the digits entered up to and

 

 

 

including the point where an impossible

 

 

 

match occurred are sent to the server

 

 

 

immediately.

 

 

 

If set to 1, give reorder tone.

 

 

 

If set to 2, allow user to accumulate digits and

 

 

 

dispatch call manually with the Send soft key.

 

 

 

 

dialplan.removeEndOfDial

0, 1

1

If set to 1, strip trailing # digit from digits sent

 

 

 

out.

 

 

 

 

This attributes also includes:

Digit Map <digitmap/>

Routing <routing/>

Digit Map <digitmap/>

A digit map is defined either by a “string” or by a list of strings. Each string in the list is an alternative numbering scheme, specified either as a set of digits or timers, or as an expression over which the gateway will attempt to find a shortest possible match.

Digit map extension letter “R” indicates that certain matched strings are replaced. The following examples shows the semantics of the syntax:

R9RRxxxxxxx—Remove 9 at the beginning of the dialed number

For example, if a customer dials 914539400, the first 9 is removed when the call is placed.

RR604Rxxxxxxx—Prepend 604 to all 7 digit numbers

For example, if a customer dials 4539400, 604 is added to the front of the number, so a call to 6044539400 is placed.

R9R604Rxxxxxxx—Replaces 9 with 604

R999R911R—Convert 999 to 911

xxR601R600Rxx—When applied on 1160122 gives 1160022

xR60xR600Rxxxxxxx—Any 60x will be replaced with 600 in the middle of the dialed number that matches

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Polycom SIP 3.1 manual This attributes also includes, Digit Map digitmap Routing routing

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.

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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.