DSP Components
511
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[VoIP Receive]-Setting
Server:
IP: The connected IP of SIP server.
RTP Port: The port that is using for
streaming.
SIP Port: The port that is using via SIP
protocol.
Codec: The format that is using for
streaming.
Buffer: The delay time of streaming.
Account:
User: The calling name(text or telephone
number) can be user-defined.
Password: The password that is using to
connected to SIP server.
RU SEC User: The calling name(text or
telephone number) of the secondary unit
when using redundant configuration.
RU SEC Password: The password of
secondary unit when using redundant
configuration using to connected to the
SIP server.
Paging: With the element of network paging,
users can multi-paging through calling.
Answer Behavior: Select the ring tone time
before the call picks up automatically. Select
manual mode to manually pick up the call by
clicking [
] Call button.
Call: Click to dial the call/receive the call.
Cancel/Hang up: Click to cancel/hang up
the call.
Telephone Book: Click to open [Contact
Manager] window and choose the contacts.
Identifier: Enter/display the SIP IP call. The
format is "[phone number]@[IP address]:[SIP
port]"
5.26.2 DTMF VoIP
Receive a telephone call and detects if a DTMF tone output logic signal responded to the key pressing
on remote telephone. It needs to work with the telephone card. Note: The [DTMF VoIP] component will
occupied with 1 x message channel, please refer to [
] chapter.
