server (e.g. Asterisk or Cisco Call Manager). Enter the IP
address in regular dot notation, e.g. 10.5.2.138.
Backup Domain (SIP) / Backup Domain 2 (SIP)
- This is the secondary (or fallback) and tertiary backup
domain. If the station loses connection to the primary SIP
domain, it will switch over to the secondary one. Enter the
IP address in regular dot notation.
Authentication User Name
- Authentication user name used to register the station to
the SIP server. Only required if the SIP server requires
authentication and is normally the same as the SIP ID.
Authentication Password
- The authentication user password used to register the
station to the SIP server. This is required only if the SIP
server requires authentication
Register interval
- Species how often the station will register, and reregister
in the SIP domain. This parameter will affect the time it
takes to detect that a connection to a SIP domain is lost.
- Enter the values in number of seconds from 60 to 999999.
The default interval is 600 seconds.
Outbound Proxy [optional]
- Enter the IP address of the outbound proxy server in
regular dot notation, e.g. 10.5.2.100
Port
- Port number used for SIP on the outbound proxy server.
The default port number is 5060.
Call Settings
Enable Auto Answer
- This is not required. Enables automatic answer after a set
number of seconds.
- Check the box to enable this function and enter the delay
in seconds in the eld for Auto Answer Delay. The
default delay setting is 0 and the maximum is 30 seconds.
Delay Call Setup
- This only applies to input buttons and DAKs. The default
delay setting is 0 and the maximum is 60 seconds.
Overlap dialing
- This will lead to the phone starting to dial each time a
digit is entered and the SIP proxy replying with ‘Number
incomplete’ until such time as the number has been
entered and the call can be initiated successfully without
the enter key having to be pressed.
DTMF method
- Choose between SIP INFO or RFC 2833 to select DTMF
signalling method.
RTP Timeout value
- This cancels a call if the station does not receive RTP
packets from the remote party. Enter values in the range
0-9999 seconds. The default setting is 0 which means
RTP timeout is disabled.
●After entering all the desired values, click Save and then
click Reboot to enable the SIP settings.
2.3 Audio Settings
To congure audio settings:
●Select SIP Conguration > Audio Settings from the
menu
Speaker Volume
- Select volume level in range 0-7. Default setting is 5
Noise Reduction Level
- The higher the noise reduction level the more
deterioration there is in audio quality.
- Default setting is 0 (i.e. the function is disabled)
Microphone Sensitivity
- Select sensitivity level in range 0-7. Default setting is 5
Remote Controlled Volume Override Mode
- This acts as simplex mode. This feature is activated after
the rst DTMF * or # is received from the remote station.
Send DTMF * to talk and DTMF # to listen.
Message Controlled Volume Override Mode
Check the box to enable the following messages:
- SIP MESSAGE “Audio_receive_only”: Turns the
microphone off and loudspeaker on
- SIP MESSAGE “Audio_send_only”: Turns microphone on
and loudspeaker off
- SIP MESSAGE “Audio_send_receive”: Turns both
microphone and loudspeaker on.
Automatic Volume Control
- Check box to enable automatic volume control that is
adjusted according to the noise level.
Debug Automatic Volume Control
- Check box to show current volume level on OLED display.
Conversation Mode
-Full Open Duplex: Normal mode with echo cancellation
-Robust Duplex: Option used when open duplex fails due
to excessive speaker loudness, microphone overload or
very high nonlinear distortions.
-Half Duplex Switching: Switches speech direction
depending on who speaks the loudest
-Push-To-Talk: Half-duplex communication. Initially the
microphone is shut off. Push the M-button to open the
microphone, and release to listen. (TCIS-1 station only)
-Open: Full Open Duplex without echo cancellation
