Polycom 3804-11530-222 manual Cfg Action Parameter Description File

Page 25

Release Notes - SIP ApplicationChanges

2.6.4 Configuration File Parameter Changes

.cfg

Action

Parameter

Description

File

 

 

 

sip

added

voIpProt.SIP.useContactInReferTo

0 = default behavior which is the same as

 

 

 

previous behavior, use URI from initial

 

 

 

call’s To header in REFER’s refer-to

 

 

 

header

 

 

 

1 = use URI from initial call’s Contact

 

 

 

header in REFER’s refer-to header when

 

 

 

setting up a transfer

sip

added

voice.gain.rx.analog.chassis.IP_330

New parameters to support SoundPoint

 

 

voice.gain.rx.analog.ringer.IP_330

IP 320 and 330 platforms which will be

 

 

voice.gain.rx.digital.chassis.IP_330

supported in a future software release. Do

 

 

voice.gain.rx.digital.ringer.IP_330

not change these values.

 

 

voice.gain.tx.analog.chassis.IP_330

 

 

 

voice.gain.tx.digital.chassis.IP_330

 

 

 

voice.rxEq.hs.IP_330.preFilter.enable

 

 

 

voice.rxEq.hs.IP_330.postFilter.enable

 

 

 

voice.rxEq.hd.IP_330.preFilter.enable

 

 

 

voice.rxEq.hd.IP_330.postFilter.enable

 

 

 

voice.rxEq.hf.IP_330.preFilter.enable

 

 

 

voice.rxEq.hf.IP_330.postFilter.enable

 

 

 

voice.txEq.hs.IP_330.preFilter.enable

 

 

 

voice.txEq.hs.IP_330.postFilter.enable

 

 

 

voice.txEq.hd.IP_330.preFilter.enable

 

 

 

voice.txEq.hd.IP_330.postFilter.enable

 

 

 

voice.txEq.hf.IP_330.preFilter.enable

 

 

 

voice.txEq.hf.IP_330.postFilter.enable

 

sip

added

voice.vad.signalAnnexB

A new line can be added to SDP

 

 

 

depending on the setting of this

 

 

 

parameter and the voice.vadEnable

 

 

 

parameter.

 

 

 

Default behavior is the same as

 

 

 

voice.vad.signalAnnexB = 0:

 

 

 

No change to the SDP

 

 

 

voice.vad.signalAnnexB = 1:

 

 

 

If voice.vadEnable=1, add attribute line

 

 

 

a=fmtp:18 annexb=”yes”

 

 

 

below a=rtpmap… attribute line (where

 

 

 

‘18’ could be replaced by another

 

 

 

payload)

 

 

 

If voice.vadEnable=0, add attribute line

 

 

 

a=fmtp:18 annexb=”no”

 

 

 

below a=rtpmap… attribute line (where

 

 

 

‘18’ could be replaced by another

 

 

 

payload)

Copyright © 2007 Polycom, Inc.

Page 19

Image 25
Contents Version SoundPoint and SoundStation IPPage Table of Contents Copyright 2007 Polycom, Inc 18.3 Reference Documents Important Notes Platform BootROM versionSystem Requirements Files Description Distribution FilesRemoved Features Added or Changed FeaturesRemove 1000 half duplex as a valid ethernet configuration VersionVersion 2.2.1 Limited Release Configuration File Parameter ChangesCfg File Action Parameter Description Cfg File Action Parameter Dhcpinform Removed support for the SoundPoint IP 300 and 500 phones Release Notes SIP Application Changes Release Notes SIP Application Changes Dhcp Cfg File Action Parameter Description Cfg File Action Parameter Description Cfg File Action Parameter Description Added logging of version information for configuration files RFC Version 2.1.1 C Following issues have been resolved with this release Release Notes SIP Application Changes Cfg Action Parameter Description File Version Removed Features Release Notes SIP Application Changes Configuration File Parameter Changes Version 2.0.3 B Following issues have been resolved with this release None Configuration File Parameter Changes Version Emergency routing is not supported on shared lines Version 2.0.1 BMalformed Rtcp packets can crash Cisco gateways Call.callWaiting.prompt has no effect Release Notes SIP Application Changes Configuration File Parameter Changes Version 2.0.0 Beta Release Only Release Notes SIP Application Changes Removed Features Release Notes SIP Application Changes Release Notes SIP Application Changes Configuration File Parameter Changes Cfg Action Parameter Description File Cfg Action Parameter Description File Version Configuration File Parameter Changes Version 1.6.6 B Version 1.6.6 C Limited DistributionAdd Support for SoundPoint IP 430 hardware platform Version Following issues have been resolved with this release Changed power reported via CDP to platform-specific values Removed Features Version Version Disabled url-dialing in main partner configuration files Contactsdirectory Cfg Action Parameter File Version 1.6.0 Beta onlyRemoved Features Configuration File Parameter Changes From Version 2.2.1 to UpgradingFrom Version 2.2.0 to From Version 2.1.2 toFrom Version 2.1.1 to 2.1.1 C From Version 2.1.1 C toFrom Version 2.1.0 to From Version 2.0.3 to 2.0.3 B From Version 2.0.3 toFrom Version 2.0.2 to From Version 2.0.0 to From Version 2.0.1 toFrom Version 1.6.7 to From Version 1.6.5 to From Version 1.6.6 toFrom Version 1.6.4 to From Version 1.6.2 to From Version 1.6.3 toFrom Version 1.6.1 to From Version 1.6.0 toNo Layer 2 QoS support for signaling protocol TCP Outstanding IssuesHttp Digest Authentication does not work on IIS Boot servers running explicit Ftps are not supportedBLA line can not place and hold more than 10 calls Active FTP mode is not supported for phone provisioningBrief audio ‘noise’ due to Srtp encryption key change Reference Documents

3804-11530-222 specifications

The Polycom 3804-11530-222 is a highly regarded communication device that exemplifies the power of modern telephony and video conferencing solutions. As part of Polycom's extensive lineup, this model is designed to enhance collaboration and productivity in various business environments.

One of the standout features of the Polycom 3804-11530-222 is its superior audio quality. The device utilizes Polycom's legendary Acoustic Clarity Technology, which ensures that every word is heard clearly, reducing background noise and echo. This is particularly crucial in conference room settings where multiple participants are involved. The technology allows for natural conversation flow, enabling users to communicate effectively without interruptions.

Another significant characteristic of the Polycom 3804-11530-222 is its robust design tailored for conference environments. The device is built to support up to six participants, making it ideal for small to medium-sized meeting rooms. Its user-friendly interface includes intuitive buttons that allow seamless operation without the need for extensive training.

The devices equipped with Polycom's HD Voice technology deliver crystal-clear audio at a full range of frequencies. This high-definition audio capability ensures that even the most nuanced tones are transmitted accurately, providing an immersive audio experience for users. Whether it's a boardroom meeting or a project discussion, the Polycom 3804-11530-222 guarantees that participants feel engaged and connected.

Additionally, the Polycom 3804-11530-222 is designed to integrate seamlessly with major unified communication platforms, including Microsoft Teams, Zoom, and Skype for Business. This compatibility makes the device a versatile addition to any corporate telecommunications setup, allowing users to transition smoothly between different communication methods.

Furthermore, the device supports multiple connectivity options, including USB and Ethernet ports. This flexibility ensures easy integration into existing infrastructures, and enables users to connect laptops and other devices quickly for presentations or video conferencing.

In summary, the Polycom 3804-11530-222 is not just a phone; it is a comprehensive communication solution that enhances collaboration and improves productivity in any professional setting. With its focus on audio clarity, user-friendly design, seamless integration, and versatile connectivity options, it continues to be a preferred choice for organizations looking to elevate their communication capabilities.