Polycom 3804-11530-222 manual Following issues have been resolved with this release

Page 50

Release Notes - SIP Application

Changes

2.17.2 Removed Features

None.

2.17.3 Corrections

The following issues have been resolved with this release:

11658: Phone continues to append to log file on FTP boot server after that file has reached its configured size limit

12613: SoundPoint IP600 and 601 phones may establish a call with no audio after holding, resuming and ending multiple calls

12949: If the phone’s first line is a shared line and cannot obtain dial tone, pressing the “NewCall” soft key does not activate the first available line

14673: Special characters such as ‘@’, ‘:’ and ‘?’ are not accepted as part of the FTP or HTTP password

14968: If the phone reboots, the app.log size can increase past the size limit

15002: If the phone’s first line is unregistered, pressing the “NewCall” soft key does not activate another line

15127: Phone may have one-way audio in a call after multiple transfers have been done

15218: If multiple contact header fields contain multiple expire values, the phone does not always pick the lowest non-zero value

15235: Phone will freeze if the SAS-VP server becomes unavailable when the phone application is starting

15339: ACK lacks the same authorization credentials as the INVITE which is a failure to comply with RFC 3261

15419: Blind transfer doesn't work for URL calling

15568: A comma in quotes in SIP address headers should be interpreted correctly

15596: Remote phone can force local conference host to resume call unexpectedly in specific scenario

15615: When a shared line call is on hold, lifting the handset seizes the last used line instead of the first available line

14939: Shared line user must press “Answer” soft key twice to answer an incoming call in some scenarios

15907: After a reboot, a phone may show "1 new missed call" which can't be cleared until another call is missed

15982: The SDP session identifier should not be changed on each re-INVITE

16021: FTP downloads may fail because incorrect timeouts are used

16141: Phone with a shared line loses hot dialed digits when remote shared line changes state, such as placing an active call on hold

Page 44

Copyright © 2007 Polycom, Inc.

Image 50
Contents SoundPoint and SoundStation IP VersionPage Table of Contents Copyright 2007 Polycom, Inc 18.3 Reference Documents System Requirements Platform BootROM versionImportant Notes Distribution Files Files DescriptionRemove 1000 half duplex as a valid ethernet configuration Added or Changed FeaturesRemoved Features VersionCfg File Action Parameter Description Configuration File Parameter ChangesVersion 2.2.1 Limited Release 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 Malformed Rtcp packets can crash Cisco gateways Version 2.0.1 BEmergency routing is not supported on shared lines 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 Add Support for SoundPoint IP 430 hardware platform Version 1.6.6 C Limited DistributionVersion 1.6.6 B 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 2.1.0 to From Version 2.1.1 C toFrom Version 2.1.1 to 2.1.1 C From Version 2.0.2 to From Version 2.0.3 toFrom Version 2.0.3 to 2.0.3 B From Version 1.6.7 to From Version 2.0.1 toFrom Version 2.0.0 to From Version 1.6.4 to From Version 1.6.6 toFrom Version 1.6.5 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.