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NETWORK SETTINGS
Use SonicIP
If set to "yes", the device will announce its IP address over the audio output during device startup.
Default: "yes"
IP Address
Enter the 4 values of the desired device IP address e.g.:
"0.0.0.0" for automatic discovery (DHCP/BOOTP, IPzator, AutoIP)
"192.168.0.12" for an internal LAN
Default: "0.0.0.0"
Netmask
Enter the 4 values of the desired Static IP e.g.:
"0.0.0.0" for a default Netmask depending on the used IP Address.
"255.255.255.0" for a C class network
Default: "255.255.255.0"
Gateway IP Address
Enter the 4 values of the desired Gateway IP address e.g.:
"0.0.0.0" for no Gateway
"192.168.0.1" for a Gateway in a LAN
Note: The Gateway has to be set only when connecting to other devices over the WAN (through a router).
Default: "0.0.0.0"
Primary DNS
In this field you can give the Exstreamer the desired primary and alternativ DNS IP address to be able to connect to URLs (e.g. www.radio.com).
Example: "195.186.1.111"
Default: "0.0.0.0"
Alternative DNS
In this field you can give the Exstreamer the desired alternativ DNS IP address in case the primary DNS is not reachable.
Example: "195.186.4.111"
Default: "0.0.0.0"
Syslog Address
Destination address for syslog messages sent by the BCL program via the SYSLOG command. Set this to your syslog logging machine, if your syslog messages are recorded centrally.
If set to 0.0.0.0, syslog messages are broadcast.
Default: "0.0.0.0"
DHCP Host Name
Name of the device sent in DHCP request. If left empty, a name based on the device's MAC address is generated automatically. Enter up to 15 Characters.
Web Server Port
Defines the port where the webserver of the Barix Exstreamer can be reached. If set to "0" the default HTTP port (80) is used.
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Default Ethernet Port
On devices with dual-ethernet connection this parameter selects the default port to
be used. In most cases selecting ETH1 is suitable, however some devices (e.g.
the PS16) provide Power-over-Ethernet (PoE) functionality on ETH2.
If you change this parameter do not forget to connect the ethernet cable to the
proper port on the rear of the device.
Default: "ETH1"
SNMP System Name
SNMP MIB entry for system name (system.sysName.0)
SNMP System Location
SNMP MIB entry for system location (system.sysLocation.0)
SNMP System Contact
SNMP MIB entry for system contact (system.sysContact.0)
This parameter can be queried using any SNMP browser but can not be updated.
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SIP PROTOCOL SETTINGS
Peer to Peer
Choose whether peer to peer calls should be allowed.
NOTE:When using P2P, the device uses always the default SIP (port 5060) and RTP (port 5004) ports. Make sure the remote peer is configured to listen on the default ports as well.
SIP Server (PBX)
Enter either the hostname/IP address of a SIP server.
SIP ID
Enter the SIP ID (username) that has been created for this device.
SIP Password
Leave this field empty if the PBX doesn't require authentication.
Listen SIP Port
Listening port for the SIP protocol messages. A value of 0 mean a default value of 5060.
Listen RTP Port
Listening port for the RTP audio blocks. A value of 0 mean a default value of 5004.
Default REGISTER Time
The value that the SIP client suggests to the SIP server when sending the REGISTER request. If this value is accepted by the SIP server, the SIP client has to register after this amount of time. Allowed values in the range of 60-3600 seconds.
NOTE: The SIP server may overwrite this value in its reply to the REGISTER request.
Default: 1800 seconds
Send NAT-Keepalives
Enable this function if the device resides behind a NAT.
Periodically Renew DNS
If enabled, it will force the SIP client to renew the DNS of the SIP server every time the registration to the server fails. Use this feature if you have a backup SIP server with the same DNS name and you wish to enable the SIP client to switch to it if the main server fails.
Registration Fail Timeout
The SIP client will automatically resolve the DNS of the SIP server if the REGISTER request fails. However, if the SIP server is down, or the REGISTER message gets lost, the SIP client will not get reply from the server. So configure here the timeout after sending a REGISTER request on the expiry of which the registration will be considered as "failed" in case of no reply from the SIP server.
NOTE: Use this feature with caution. Setting this value too low may result in registration malfunctioning. If unsure, leave it to the default value.
Default: 10 seconds
Call Timeout
Enter here in minutes (1-255) the maximum duration of the call. After the expiry of this timeout the call will be unconditionally closed.
Default: 0 (disabled)
Debug Mode
Sends received or sent SIP messages through Syslog.
CIP Port Number
Set here the port to listen to, and to send CIP control messages.
Default: 41794
SIP Groups
Set here a comma delimited list of groups in which this device is a member of.
NOTE: All devices must be members of the group "ALL", so do not delete it from here.
SIP Display ID
Set here a name you will like to be displayed on the remote SIP client when calling.
comma delimited list of groups in which this device is a member of.
NOTE: This field can be used only from CIP capable SIP clients.
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OUTBOUND CALL SETTINGS
Auto Hangup Time
If enabled, the device will automatically hang-up after the configured amount of time if no one answers the call. This option is suitable for door station panels that have only one ring button, and no button to cancel the initiated call.
Values: 5 to 90 seconds. Default: 0 (disabled)
NOTE: When the Auto Hangup Time is disabled, a second press on the Input0/Input1 button will cancel the started call setup. Please do not disable it if the device is intended to be used as doorstation. If the call is not cancelled with a second button press, the SIP client will anyway unconditionally drop the call after 120 seconds.
Call on Level
If enabled, call can be initiated by audio level. If set to "Yes", then Call on Level ID, Level Threshold and Close Call on Level options are also visible.
NOTE: Call on level is unsupported with "Background Music" enabled.
Call on Level ID
SIP extension of the device that will be called on audio level detection.
In case of using "Peer to Peer" mode, enter here the ID and the IP address of the remote peer, for example: 1234@192.168.0.123
Level Threshold
Minimal audio level to initiate a call. If the input audio level reaches at least the configured threshold, a call wiil be initiated. The same threshold level is used to terminate the call if the Close Call on Level option is enabled
Values: 0 to 32767. Default: 1000
Close Call on Level
When enabled, and the input peak level is below the level threshold for the configured amount of time, the call is terminated.
Values: 5 to 30 seconds. Default: 0 (disabled)
Beep On Call Answer
If enabled, the SIP client will play a short beep sound when the remote party answers the call.
Default: Off
Call on Device Inputs
Extensions to be called when the correspondent input is closed. The available number of inputs depends from the Barix hardware model.
In case of using "Peer to Peer" mode, enter here the ID and the IP address of the remote peers, for example: 1234@192.168.0.123
NOTE 1:For calling SIP clients that support the CIP protocol, only the ID is required. The IP address will be provided by the CIP protocol.
NOTE 2: To initiate a group call, put here the word "CALL:" followed by the comma separated list (no spaces allowed) of groups to call. For example, "CALL:OFFICE,HOME" will start a group call to all devices that are members of groups OFFICE and HOME.
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INBOUND CALLS
Input Buffer Level
Maximum delay of the input audio buffer in milliseconds. Decrease this value to minimize delay, increase this value to prevent audio dropouts.
Default setting is "300" ms.
Phone Pickup Mode
Choose how calls should be answered.
autoanswer : the call is auto-answered.
autopick up after timeout : in this mode the bell button is used to answer and terminate a call. If the call is not answered inside "Pick/Hang up After" time period, the device will answer.
autohang up after timeout : (default) in this mode the bell button is used to answer and terminate the call. If the call is not answered inside "Pick/hang up after" time period, the device will decline the call.
not callable : in this mode incoming calls will be declined.
Pick/Hang up After
Set pick/hang up delay if no answer. Only active if the phone pickup mode is set to manual.
Stream Timeout
In some scenarios the remote party may go offline without explicitly closing the active call. In this case the SIP client may stay in active call state for unlimited amount of time. Set here the time in minutes after which the SIP client will close the active call if there is no audio stream received. You can change between 0 and 600 minutes.
NOTE: Please have in mind that this setting is ignored and reset to "disabled" if you select SIP Monitoring Point profile.
Default setting is "0 min (disabled)"
Beep On Call Answer
If enabled, the SIP client will play a short beep sound when the incoming call is answered.
Default: Off
Door Open Code
Set the secret code to open the door by entering this DTMF sequence while communicating with the device.
When the sequence is detected from the remote peer, the first relay contact is closed.
Leave it empty to disable this functionality.
Open Door Relay for
Set the contact closure duration. After this period, the relay contact is switched back to open.
Enable Relay
Select here when you would like to have the relay enabled.
Default setting is "on door open command"
NOTE: The status of the relay can still be modified with a DTMF command.
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AUDIO SETTINGS
Input Source
Select the desired input source.
Default setting is "Mic".
Encoding
Choose between "uLaw", "aLaw", or G.722.
Note: The G.722 selection option is visible only if supported by the HW.
Default setting is "uLaw".
Volume
Choose between "0%" and "100%" in 5% steps.
Default: "50%"
Microphone Gain
Microphone gain dB, increase if your microphone is too faint, decrease if it's too loud or overdriven.
Default: "21dB"
A/D Amplifier Gain
A/D converter preamplification in dB. Increase if the audio signal too faint, decrease if it's too loud or overdriven.
Default: "0dB"
Talk Mode
Choose between Fullduplex and Halfduplex talk mode. This option is available only in SIP Door Station profile. When selected, the user has also the possibility to enable the support for AI Phone door panels.
Default: "FDX" at factory defaults, and "HDX" when the SIP Door Station Profile is applied
Output Trigger Level
Defines the audio level from the incoming audio RTP stream which will trigger the SIP client in "listen" mode. This setting is ignored if the door station is configured in FDX mode.
Values: 0 to 32767.
Default: "1000"
Trigger Level Timeout
Defines the timeout in milliseconds, after which if no audio is detected from the incoming RTP stream, the SIP Client should switch back to "talk" mode. This setting is ignored if the door station is configured in FDX mode.
Default: "200 ms"
Acoustic Echo Cancellation
Enable or disable the AEC here.
Note: The AEC option is visible only if supported by the HW.
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CONTROL INTERFACES
Use Serial Port For
These settings adjust the purpose of the serial port. You can use between the following values:
No Use: In this mode the serial port is not used.
Serial Control Interface: In this mode the serial port is used to send commands to the SIP client. For more information how to use it please refer to the SIP User Manual.
Default: "NO Use"
Baud Rate
Select the serial transmission speed ("300" to "230400" Baud).
Default: "9600"
Data Bits
Select "7" or "8" data bits.
Default: "8"
Parity
Select "no", "even" or "odd" parity.
Default: "no"
Stop Bits
Select "1" or "2" stop bits.
Default: "1"
Flow Control
Select the type of flow control:
RTS/CTS signals not used: "none"
RS232: "Software flow control(XON/XOFF)" or "Hardware flow control (RTS/CTS)"
RS485: "RS485 direction control"
Default: "none"
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SECURITY SETTINGS
Reset Function
Enable or disable the "Reset" function on the Reset button and on the WEB UI. In order to restart the device press the Reset button once.
Default: "enabled"
Factory Defaults
Enable or disable the "Factory Defaults" function on the Reset button. In order to revert all settings to factory defaults keep the Reset button pressed until the red LED starts blinking (approx. 10 seconds).
Default: "enabled"
Update Function
Enable or disable the WEB Update function of the device. If the Update function is disabled, the only way to update the firmware is to use the serial rescue.
Default: "enabled"
Set Password
This is visible as long as no password is set.
Enter a password (up to 25 characters) and hit the "Apply" button. After the restart you should close the browser window and open a new browser window. You will be asked to supply user name and password. The user name can be omitted but the password has to be supplied in order to see the web configuration.
Old Password / New Password
These fields are visible as long as a password is set.
To allow free access (clearing the password) enter the old password and leave the field "New Password" empty, then hit the "Apply" button.
After the restart you will not be asked for user name and password anymore.
To change the password enter the old password and enter the new password in the field "New Password", then hit the "Apply" button.
After the restart you will be asked for user name and password. The user name can be omitted but the new password has to be supplied in order to see the web configuration.
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