Digital Acoustics TalkMaster FOCUS, Reference Manual

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IP Endpoints Tab
47
If the Primary SIP Server or Registrar becomes unavailable, the IP Endpoint will automatically
register with the Secondary Registrar and Server. The IP Endpoint will automatically de-register
with the Secondary Registrar and register with the Primary Registrar when it is available

Secondary Registrar - Specifies the IP Address or DNS name of the secondary SIP Registrar.
Typically, the SIP Registrar and the SIP Server are the same

Secondary SIP Server - Specifies the IP Address or DNS name of the secondary SIP Server
SIP Port Information

SIP Local Port - Specifies the IP Endpoint's port to be used for the SIP protocol. This defaults to
port 5060

RTP Port - Specifies the IP Endpoint's port to be used for the RTP traffic. This defaults to port
46000
SIP Extension Information

SIP Extension - Specify the SIP Extension that has been assigned in the SIP Server

SIP Password - Specify the SIP Password that has been assigned in the SIP Server

SIP Authentication - Specify the User associated with the Extension. Typically, this is the same
as the SIP Extension
Processing Options

SIP Full Duplex - Check this box if the IP7 Endpoint will be used in Full Duplex mode
Some IP7s are capable of utilizing full duplex audio, but only the IP7-FD model supports
Acoustic Echo Cancellation (AEC). AEC is required in installations where the audio coming from
the IP Endpoint's Speaker can be picked up by the IP Endpoint's Microphone. To use Full Duplex
with older IP7's, a handset or headset is required to prevent an echo from being heard by the
device calling the IP7
If this option is not checked, calls will be in half-duplex mode. The phone operator will be able
to speak and the user at the IP Endpoint will be able to listen when the call is initiated. The phone
operator can press 0 on the phone to toggle the direction of speaking and listening

Out Dial on Queue Overflow - This option is only available when the Type field on the General--
> Configuration tab is set to Client and the eSIP Endpoint is configured for SIP Extension
Only . This option is not available when the Dial a SIP Extension on PTT has been selected. If
this option is checked and the IP Endpoint's Talk button is pressed, the call will be sent to an
Operator Console. If the Queue that the Endpoint is assigned to goes into Overflow status, the
incoming call will be removed from the Operator Console and the endpoint will use the number
from the
Out Dial Extension field to dial the SIP server.

Out Dial Extension - If the Dial a SIP Extension on PTT option is selected, the number listed
here will be dialed automatically when the IP7's Talk button is pressed . If the Out Dial on
Queue Overflow option is selected and a Queue the IP Endpoint is a assigned to overflows, the
TalkMaster FOCUS Server will instruct the IP Endpoint to dial this number

Door Code - Specifies the keys on the Phone's keypad to be pressed for the IP Endpoint to
activate its Relay to open a door. If more that 1 digit is entered, the # key must be entered in
addition to the Door Code through the Phones keypad

Call Timeout - While a call is in progress, specifies the number of seconds that audio can drop
out before the call will be automatically terminated