Configuration Files

This configuration attribute is defined as follows:

 

Permitted

 

 

Attribute

Values

Default

Interpretation

 

 

 

 

nat.ip

dotted-decima

Null

IP address to advertise within SIP signaling - should

 

l IP address

 

match the external IP address used by the NAT device.

 

 

 

 

nat.signalPort

1024 to 65535

Null

If non-Null, this port will be used by the phone for SIP

 

 

 

signaling, overriding the value set for

 

 

 

voIpProt.local.signalPort in sip.cfg.

nat.mediaPortStart

1024 to 65535

Null

If non-Null, this attribute will be used to set the initially

 

 

 

allocated RTP port, overriding the value set for

 

 

 

tcpIpApp.port.rtp.mediaPortRangeStart in sip.cfg.

 

 

 

Refer to RTP <rtp/> on page A-62.

 

 

 

 

nat.keepalive.interval

0 to 3600

Null

If non-Null (or 0), the keepalive interval in seconds. This

 

 

 

parameter is used to set the interval at which phones will

 

 

 

send a keep-alive packet to the gateway/NAT device to

 

 

 

keep the communication port open so that NAT can

 

 

 

continue to function as setup initially.

 

 

 

The Microsoft Live Communications Server 2005

 

 

 

keepalive feature will override this interval. If you want to

 

 

 

deploy phones behind a NAT and connect them to Live

 

 

 

Communications Server, the keepalive interval received

 

 

 

from the Live Communications Server must be short

 

 

 

enough to keep the NAT port open. Once the TCP

 

 

 

connection is closed, the phones stop sending keep-alive

 

 

 

packets.

 

 

 

 

Attendant <attendant/>

Note

These attributes are available on SoundPoint IP 320/330, 430, 550, 560, 600, 601,

 

650, and 670 phones only.

 

 

The Busy Lamp Field (BLF) / attendant console feature enhances support for a phone-based attendant console.

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Polycom SIP 3.1 manual Attendant attendant, VoIpProt.local.signalPort in sip.cfg

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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