Local Contact Directory

Configuring Your System

IP 4000, 6000, and 7000 conference phones are full-duplex speakerphones. The SoundPoint IP 301 phones is a listen-only speakerphone. The SoundPoint IP phones provide dedicated keys for convenient selection of either the speakerphone or headset.

Only the SoundPoint IP 320, 330, 430, 550, 560, 650, and 670 desktop phones can be configured to use the electronic hookswitch. For more information, refer to “Technical Bulletin 35150: Using an Electronic Hookswitch with SountPoint IP Phones“at http://www.polycom.com/usa/en/support/voice/soundpoint_ip/VoIP_T echnical_Bulletins_pub.html .

Configuration changes can performed centrally at the boot server or locally:

Central	Configuration file:	Enable or disable persistent headset mode.
(boot server)	sip.cfg	For more information, refer to User Preferences <up/> on page A-25.
		Enable or disable hands-free speakerphone mode.
		• For more information, refer to User Preferences <up/> on page
		A-25.

	Configuration file:	Specify whether or not the electronic hookswitch is enabled and what
	phone1.cfg	type of headset is attached.
		• For more information, refer to User Preferences
		<user_preferences/>on page A-107.

Local	Web Server	Enable or disable persistent headset mode.
	(if enabled)	Navigate to: http://<phoneIPAddress>/coreConf.htm#us

	Local Phone User	Enable or disable persistent headset mode through the Settings
	Interface	menu (Settings > Basic > Preferences > Headset > Headet
		Memory Mode).
		Enable or disable hands-free speakerphone mode through the
		Settings menu (Settings > Advanced > Admin Settings > Phone
		Settings).
		Changes are saved to local flash and backed up to <Ethernet
		address>-phone.cfgon the boot server. Changes will permanently
		override global settings unless deleted through the Reset Local
		Config menu selection and the <Ethernet address>-phone.cfgis
		removed from the boot server.

The phone maintains a local contact directory. The directory can be downloaded from the boot server and edited locally (if configured in that way). Contact information from previous calls may be easily added to the directory for convenient future access.

The directory is the central database for several other features including speed-dial, distinctive incoming call treatment, presence, and instant messaging.

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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.

One of the standout features of Polycom SIP 3.1 is its robust interoperability. This protocol supports a wide range of endpoints and platforms, allowing organizations to integrate their existing systems with new technologies effortlessly. This flexibility ensures that businesses can leverage their previous investments while upgrading to the latest communication tools.

Another key aspect of Polycom SIP 3.1 is its enhanced call management capabilities. The protocol facilitates efficient call handling, enabling users to manage multiple calls seamlessly. Features like call hold, transfer, and conferencing are streamlined, which enhances productivity by allowing for efficient collaboration among team members. Additionally, it is optimized for low latency and high-quality audio, ensuring that conversations are clear and free from disruptions.

Security is paramount in today’s digital landscape, and Polycom SIP 3.1 addresses this concern with advanced encryption standards. By integrating security features such as Transport Layer Security (TLS) and Secure Real-time Transport Protocol (SRTP), it protects sensitive communication from unauthorized access and ensures that data remains confidential throughout the call.

Polycom SIP 3.1 also boasts compatibility with various video codecs, making it a versatile choice for video conferencing. This compatibility ensures high-quality video streams, which is essential for effective visual communication in remote meetings. Furthermore, the support for the H.264 codec provides efficient bandwidth usage, making high-definition video conferencing accessible, even in varying network conditions.

Moreover, the protocol provides strong support for presence and instant messaging, which enhances real-time communication among users. This integration of voice, video, and messaging capabilities fosters a more connected and collaborative work environment, allowing teams to engage effectively regardless of their geographical locations.

In summary, Polycom SIP 3.1 stands out as a sophisticated solution tailored to meet the demands of modern business communication. With its emphasis on interoperability, call management, security, video quality, and real-time collaboration, it caters to companies of all sizes seeking to optimize their communication infrastructure and enhance productivity in the workplace. As businesses continue to navigate the complexities of digital communication, Polycom SIP 3.1 remains a compelling choice in the market.

<user_preferences/>on page A-107.

Settings).

Local Contact Directory

SIP 3.1 specifications