Polycom 3804-11530-222 manual Configuration File Parameter Changes

Page 43

Release Notes - SIP Application

Changes

15679: Ring Type 12 (Ringback-style) sounds incomplete after the first ring

15694: Phone crashes and reboots when 'Exit' is pressed from Network

Configuration menu in Korean Language

15730: If a menu is displayed when a call is missed on the SoundPoint IP 300 and 301 phones, the missed call count is not updated on the idle display

15766: Display is incorrect after selecting name dialing then entering and exiting a call list while dial tone is playing

15781: After putting a local conference on hold then splitting the calls then joining them, the first call may remain on hold

15855: In the Instant Msg menu of the SoundPoint IP 300 and 301 phones, "x/Ascii" is not displayed after pressing the "1/A/a" softkey

2.13.4 Configuration File Parameter Changes

.cfg

Action

Parameter

Description

File

 

 

 

sip

added

voIpProt.server.x.expires.overlap

The number of seconds before the

 

 

 

expiration time returned by server ‘x’ at

 

 

 

which the phone should try to re-register.

 

 

 

The phone will try to re-register at half the

 

 

 

expiration time returned by the server if that

 

 

 

value is less than the configured overlap

 

 

 

value.

 

 

 

Default = 60. Minimum = 5, maximum =

 

 

 

65535.

sip

added

voIpProt.SIP.ms-forking

Default = 0. Can be 0 or 1.

 

 

 

0 = Support for MS-forking is disabled.

 

 

 

1 = Support for MS-forking is enabled and

 

 

 

the phone will reject all Instant Message

 

 

 

INVITEs. This parameter is relevant for LCS

 

 

 

server installations.

 

 

 

Note that if any endpoint registered to the

 

 

 

same account has MS-forking

 

 

 

disabled, all other endpoints default back to

 

 

 

non-forking mode. Windows Messenger

 

 

 

does not use MS-forking so be aware of this

 

 

 

behavior if one of the endpoints is Windows

 

 

 

Messenger.

sip

added

voIpProt.SIP.dialog.usePvalue

Default = 0. Can be 0 or 1.

 

 

 

0 = Phone uses “pval” field name in Dialog.

 

 

 

This obeys the draft-ietf-sipping-dialog-

 

 

 

package-06.txt draft.

 

 

 

1 = Phone uses a field name of “pvalue”.

sip

added

voIpProt.SIP.connectionReuse.useAli

Default = 0. Can be 0 or 1.

 

 

as

0 = old behaviour

 

 

 

1 = Phone uses the connection reuse draft

 

 

 

which introduces "alias".

sip

added

se.pat.callProg.15.name="secondary

Same configuration method as primary dial

 

 

dial"

tone. Allows a different tone to be

 

 

se.pat.callProg.15.inst.1.type="chord"

configured for secondary dial tone.

 

 

se.pat.callProg.15.inst.1.value="1"

 

Copyright © 2007 Polycom, Inc.

Page 37

Image 43
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 FilesVersion Added or Changed FeaturesRemoved Features Remove 1000 half duplex as a valid ethernet configurationVersion 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 2.1.2 to UpgradingFrom Version 2.2.1 to From Version 2.2.0 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 1.6.0 to From Version 1.6.3 toFrom Version 1.6.2 to From Version 1.6.1 toBoot servers running explicit Ftps are not supported Outstanding IssuesNo Layer 2 QoS support for signaling protocol TCP Http Digest Authentication does not work on IISBLA 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.