Polycom 3804-11530-222 manual Version 2.0.3 B, Added or Changed Features

Page 30

 

Release Notes - SIP Application

Changes

 

 

 

 

 

 

 

.cfg

Action

Parameter

Description

 

File

 

 

 

 

 

sip

added

voIpProt.SIP.useSendonlyHold

Can be set to 0 or 1. Null default is 0.

 

 

 

 

 

Default in sip.cfg is 1.

 

 

 

 

 

If set to 1, the phone will send a reinvite

 

 

 

 

 

with a stream mode attribute of “sendonly”

 

 

 

 

 

when a call is put on hold. This is the

 

 

 

 

 

same as the previous behavior.

 

 

 

 

 

If set to 0, the phone will send a reinvite

 

 

 

 

 

with a stream mode attribute of “inactive”

 

 

 

 

 

when a call is put on hold.

 

 

 

 

 

Note:

 

 

 

 

 

The phone will ignore the value of this

 

 

 

 

 

parameter if set to 1 when the parameter

 

 

 

 

 

voIpProt.SIP.useRFC2543hold

 

 

 

 

 

is also set to 1 (default is 0).

 

 

sip

added

dialplan.applyToUserSend="1"

Refer to Technical Bulletin 11572.

 

 

 

 

dialplan.applyToUserDial="1"

 

 

 

 

 

dialplan.applyToCallListDial="0"

 

 

 

 

 

dialplan.applyToDirectoryDial="0"

 

 

 

sip

changed

dialplan.digitmap.timeOut="3" to

Refer to Technical Bulletin 11572.

 

 

 

 

"333333"

 

 

 

sip

changed

tcpIpApp.sntp.daylightSavings.start.mo

Changes to support new daylight savings

 

 

 

 

nth="4" to “3”

time rules.

 

 

sip

changed

tcpIpApp.sntp.daylightSavings.start.dat

 

 

 

 

 

e="1" to “8”

 

 

 

sip

changed

tcpIpApp.sntp.daylightSavings.stop.mon

 

 

 

 

 

th="10" to “11”

 

 

 

sip

changed

tcpIpApp.sntp.daylightSavings.stop.day

 

 

 

 

 

OfWeek.lastInMonth="1" to “0”

 

 

 

sip

added

call.stickyAutoLineSeize.onHookDialing

Refer to Administrator’s Guide Addendum

 

 

 

 

 

for SIP 2.1.

 

 

sip

changed

voice.gain.rx.digital.chassis.IP_650="-9"

Gain changes required to match new

 

 

 

 

to “6”

software load.

 

 

sip

changed

voice.gain.rx.digital.ringer.IP_650="-21"

 

 

 

 

 

to “-12”

 

 

 

sip

changed

voice.handset.sidetone.adjust.IP_430="

 

 

 

 

 

-12" to “-13”

 

 

 

sip

added

voIpProt.server.x.transport and

Added “TCPOnly” as a possible value for

 

 

 

 

voIpProt.SIP.outboundProxy.transport

these existing parameters.

 

2.8 Version 2.0.3 B

2.8.1 Added or Changed Features

14874: Added support for SoundPoint IP 650 platform

15775: Added support for LCD backlight on SoundPoint IP 650

15852: Added support for 32 MB of memory on SoundPoint IP 650

15853: Added support for G.722 audio code on SoundPoint IP 650

16335: Added support for 8 MB of flash on SoundPoint IP 650

Page 24

Copyright © 2007 Polycom, Inc.

Image 30
Contents SoundPoint and SoundStation IP VersionPage Table of Contents Copyright 2007 Polycom, Inc 18.3 Reference Documents Platform BootROM version Important NotesSystem Requirements Distribution Files Files DescriptionRemove 1000 half duplex as a valid ethernet configuration Added or Changed FeaturesRemoved Features VersionConfiguration File Parameter Changes Version 2.2.1 Limited ReleaseCfg 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 Version 2.0.1 B Emergency routing is not supported on shared linesMalformed 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 C Limited Distribution Version 1.6.6 BAdd 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 Version 1.6.0 Beta only Cfg Action Parameter FileRemoved Features Configuration File Parameter Changes From Version 2.2.0 to UpgradingFrom Version 2.2.1 to From Version 2.1.2 toFrom Version 2.1.1 C to From Version 2.1.1 to 2.1.1 CFrom Version 2.1.0 to From Version 2.0.3 to From Version 2.0.3 to 2.0.3 BFrom Version 2.0.2 to From Version 2.0.1 to From Version 2.0.0 toFrom Version 1.6.7 to From Version 1.6.6 to From Version 1.6.5 toFrom Version 1.6.4 to From Version 1.6.1 to From Version 1.6.3 toFrom Version 1.6.2 to From Version 1.6.0 toHttp Digest Authentication does not work on IIS Outstanding IssuesNo Layer 2 QoS support for signaling protocol TCP Boot servers running explicit Ftps are not supportedActive FTP mode is not supported for phone provisioning BLA line can not place and hold more than 10 callsReference Documents Brief audio ‘noise’ due to Srtp encryption key change

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.