Polycom SIP 3.1 manual Connected Party Identification, Context Sensitive Volume Control

Models: SIP 3.1

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<serverMissedCall/> on page A-112.

 

 

Configuring Your System

 

 

 

 

Configuration changes can performed centrally at the boot server:

 

 

 

Central

Configuration file:

Turn this feature on or off.

(boot server)

sip.cfg

For more information, refer to Feature <feature/> on page A-92.

 

 

 

 

Configuration file:

Specify per-registration whether all missed-call events or only

 

phone1.cfg

remote/server-generated missed-call events will be displayed.

 

 

For more information, refer to Missed Call Configuration

 

 

<serverMissedCall/> on page A-112.

 

 

 

Connected Party Identification

The identity of the remote party to which the user has connected is displayed and logged, if the name and ID is provided by the call server. The connected party identity is derived from the network signaling. In some cases the remote party will be different from the called party identity due to network call diversion.

There are no related configuration changes.

Context Sensitive Volume Control

The volume of user interface sound effects, such as the ringer, and the receive volume of call audio is adjustable. While transmit levels are fixed according to the TIA/EIA-810-A standard, receive volume is adjustable. For SoundPoint IP and phones, if using the default configuration parameters, the receive handset/headset volume resets to nominal after each call to comply with regulatory requirements. Handsfree volume persists with subsequent calls.

Configuration changes can performed centrally at the boot server:

Central (boot server)

Configuration file:

sip.cfg

Adjust receive and handset/headset volume.

For more information, refer to Volume Persistence <volume/> on page A-42.

Customizable Audio Sound Effects

Audio sound effects used for incoming call alerting and other indications are customizable. Sound effects can be composed of patterns of synthesized tones or sample audio files. The default sample audio files may be replaced with alternates in .wav file format. Supported .wav formats include:

mono G.711 (13-bit dynamic range, 8-khz sample rate)

mono L16/16000 (16-bit dynamic range, 16-kHz sample rate)

mono L16/32000 (16-bit dynamic range, 32-kHz sample rate)

mono L16/48000 (16-bit dynamic range, 48-kHz sample rate)

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Page 59
Image 59
Polycom SIP 3.1 manual Connected Party Identification, Context Sensitive Volume Control, Customizable Audio Sound Effects