Administrator’s Guide SoundPoint IP / SoundStation IP

Voice Mail Integration

The phone is compatible with voice mail servers. The subscribe contact and callback mode can be configured per user/registration on the phone. The phone can be configured with a SIP URL to be called automatically by the phone when the user elects to retrieve messages. Voice mail access can be configured to be through a single key press (for example, the Messages key on the SoundPoint IP 430, 500, 501, 550, 560, 600, 601, 650, and 670). A message-waiting signal from a voice mail server triggers the message-waiting indicator to flash and the call waiting audio tone is played through the active audio path.

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

Central

Configuration file:

For one-touch voice mail access, enable the “one-touch voice mail”

(boot server)

sip.cfg

user preference.

 

 

For more information, refer to User Preferences <up/> on page

 

 

A-25.

 

 

 

 

Configuration file:

For one-touch voice mail access, bypass instant messages to remove

 

phone1.cfg

the step of selecting between instant messages and voice mail after

 

 

pressing the Messages key on the SoundPoint IP 430, 500, 501, 550,

 

 

560, 600, 601, 650, and 670 (Instant messages are still accessible

 

 

from the Main Menu).

 

 

On a per-registration basis, specify a subscribe contact for solicited

 

 

NOTIFY applications, a callback mode (self call-back or another

 

 

contact), and the contact to call when the user accesses voice mail.

 

 

For more information, refer to Messaging <msg/> on page A-119.

 

 

 

Local

Web Server

For one-touch voice mail access, enable the “one-touch voice mail”

 

(if enabled)

user preference and bypass instant messages to remove the step of

 

selecting between instant messages and voice mail after pressing the

 

 

 

 

Messages key on the SoundPoint IP 430, 500, 501, 550, 560, 600,

 

 

601, 650, and 670 (Instant messages are still accessible from the

 

 

Main Menu).

 

 

Navigate to http://<phoneIPAddress>/coreConf.htm#us

 

 

On a per-registration basis, specify a subscribe contact for solicited

 

 

NOTIFY applications, a callback mode (self call-back or another

 

 

contact) to call when the user accesses voice mail.

 

 

Navigate to http://<phoneIPAddress>/reg.htm

 

 

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.cfg is

 

 

removed from the boot server.

 

 

 

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Polycom SIP 3.1 manual Voice Mail Integration, Phone1.cfg

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.