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SIP ACCOUNT SETTINGS

SIP Server (PBX)
Enter either the hostname/IP address of a SIP server. For example using the style: sip.server.com

SIP User Name
Enter the Username (SIP ID) that has been created for this device on the SIP server.

Auth User Name
Some SIP servers do require Authorization User Name in addition to the SIP ID. If your SIP server is not using Auth User Name, leave it empty, otherwise the SIP authentication may fail.

SIP/Auth User Password
Enter the Auth User (when set), or SIP User (SIP ID) password that is used to authenticate to the PBX.

Outbound Proxy
Some SIP servers do require the communication to be done always via outbound SIP proxy. If this is the case for your SIP server, enter the proxy URL/IP address here.

Registration Timeout
Allows changing the time period for the registration of the SIP client. Allowed values are between 60s and 3600s.
Note:The SIP server might override this value.
Default: "600ms"

RTP Stream Timeout
The SIP Client will automatically close the active call if the incoming RTP stream from the remote peer is missing for the configured amount of time. Allowed values are between 0 (disabled) and 3600 seconds.
Note:The call might be dropped, when this option is enabled, and the remote peer has a speech silence detector that is configured to stop sending RTP stream instead of sending a stream with silence.
Default: "10s"

Autodetect NAT
Allows changing contact header if necessary to work with symmetric NAT.
Default: "On"

Autoanswer Incoming Call
If set, the SIP client will answer automatically the incoming call.
Default: "Off"

Play Beep on Call Answer
If set, the SIP client will play a beep tone:
  • Disabled: No beep tone will be played
  • Play on remote peer: The beep tone will be sent to the remote peer
  • Play also locally: The beep tone will be sent to the remote peer, and also played locally
Default: "Off"

Autodial
If set, the SIP client will automatically dial the peer configured as "Call On Input ID" when the it starts.
NOTE: Use the "On call close/drop" with caution! When set, the SIP client will try to redial again the peer configured as "Call On Input ID". You may end up in a sitiation where the only way to really stop the call is to power off the device, or change the "Autodial" option to disabled!
Default: "Disabled"

Call On Input ID
Configure here the SIP number/SIP URI to be dialed by default with button "Dial" or automatically when the "Autodial On Start" option is set.
NOTE: The Input ID must be in the format "sip_number@sip.server.com". The "sip:" is added automatically.
Default: "change_me"

DTMF Pattern
Configure here the DTMF digit sequence to toggle the RTS output of the serial port.
NOTE: You can use the RTS output to trigger external relay.

SIP CODECS SETTINGS
Enable or disable in this section all the SIP audio codecs, supported by theSIP client
Default: "All On"

SIMPLE PLAYER SETTINGS

Play MP3 Files/Stream
Select here whether you like the Player to play MP3 files from the configured Media Folder, or to play RTP/HTTP stream formats
Default: "Stream"

Play MP3 Files/Stream
Configure here the folder on the SD card where you have the MP3 files stored. Use SCP to copy them to the SD card.
NOTE: You do not need to create a playlist file. The player creates automatically a playlist of all MP3 files it finds in this folder, and then plays it.
Default: "simple_player"

Stream Url
Configure here the URL of the stream you would like to play. You can include also the port number (ex. rtp://239.0.3.4:12345)
Default: "http://www.barix.com/radio.m3u"

RTP Stream Buffer
Sets the RTP input buffer to be used for the playback of RTP streams in the range of 10-1000 ms.
NOTE: For playing HTTP streams the player uses a fixed buffer of 10 seconds.
Default: "300 ms"

AUDIO SETTINGS

Audio Input/Output Device
Allows to select between the internal audio card, and the plugged USB sound card (if detected).
Note:Please reload the configuration page after plugging the USB card in order to be shown in the selection drop-down list.
Default: "Internal Sound Card"

INTERNAL AUDIO CARD SETTINGS

Volume
Adjusts the Audio volume
Default: "50%"

Mic/Line-in
Select here whether you use the audio input for Microphone or Line In
Default: "Mic"

Microphone Gain
Adjusts the microphone sensitivity within the range from 8 to 55.25 dB
Default: "20 dB"

A/D Amplifier Gain
Adjusts the Line In sensitivity within the range from -12 to +21 dB
Default: "0 dB"

AEC
Enables or disables the use of Acoustic Echo Cancellation (AEC). Select explicitly SPEEX if required to use AEC.
Default: "Disabled"

AEC tail
Sets the buffer length of the incoming audio, over which the SIP client shall calculate the parameters of the used Acoustic Echo Cancellation (AEC) algorithm. Note:You can set it between 1 and 1000 ms. Setting the AEC to "Disabled" sets the AEC tail to 0 in the SIP application, and hides the "AEC tail" option from the configuration page.
Default: "250 ms"

USB AUDIO SETTINGS

In case there is any USB audio card detected, this section will be populated automatically with the settings of the controls, provided by its USB audio driver.

SIP AUDIO SETTINGS

VAD
Controls the Voice Activity Detection. When activated, the VAD that the speaker is not talking, and starts sending complete silence. This may cause the remote party to wrongly assume that the communication is broken, or that the call has dropped.
Default: "Disabled"

Audio Mode
Switches between "Full Duplex" and "Half Duplex" mode. When "Half Duplex" mode is activated, the local audio input/MIC is muted while the remote speaker is talking (switches to "Listen Mode") to avoid creating acoustic loopback when used with HDX door panels, or systems. As soon as the remote speaker stops talking, the audio input/MIC is enabled, and the output audio is muted (switches to "Talk Mode").
Default: "Full Duplex"

HDX Trigger Level
Configure here the audio level of the remote speaker, above which the device will switch to "Listen Mode". Allowed range: 0 to -60dB
Default: "-40dB"

HDX Trigger Timeout
Configure here the time with remote audio level below the HDX Trigger Level, after which the device will switch back to "Talk Mode". Allowed range: 0 to 2000ms
Default: "250ms"

Mic/Line In Latency
Set the sound input buffer size/latency in msec. Setting the value to lower value will lower the latency but may affect the sound stability.
Default: "100ms"

Playback Latency
Set the sound output buffer size/latency in msec. Setting the value to lower value will lower the latency but may affect the sound stability.
Default: "100ms"

Audio Channels
Open both sound device and the conference bridge in stereo or mono mode. Left and right channels may be mixed if the media port registered to the conference bridge is not stereo.
Default: "Mono"

NETWORK SETTINGS

Use Avahi discovery
If set to "yes", the Avahi discovery daemon is actived.
This daemon implements zero-configuration networking, including a system for multicast DNS/DNS-SD service discovery
Default: "no"

Use SonicIP®
If set to "yes", the device will announce its IP address over the audio output.
Default: "yes"

SonicIP® Volume
Sets the volume at which the SonicIP® will be announced at boot.
Default: "50 %"

Protocol
Select "DHCP" for automatic assignment of IP address, Netmask, Gateway and Primary/Alternative DNS.
Select "Static" for manually assigning IP address, Netmask, Gateway and Primary/Alternative DNS.
Note: If set to "Static" all fields are mandatory (except Alternative DNS and Syslog Address).
Default: "DHCP"

DHCP Host Name
When "DHCP" is selected as protocol, you can optionally set a DHCP Host Name to be sent in the DHCP request.

Web server port
Defines the port where the webserver of the Barix Store&Play Device can be reached.
Default: "80"

Syslog Address
Enter the destination IP address for Syslog messages sent by the SYSLOG command.
Note: The Syslog messages are not broadcasted, so you need to set explicitly the IP address of your Syslog logging device in order to be able to receive the Syslog messages.
Disable: "0.0.0.0 (disabled)"
Default: ""

TIME SETTINGS

Allows to set up to 3 NTP servers for internet time adjustment.

SNMP SETTINGS

Enable SNMP
Enable or disable the SNMP functionality.
Default: "Off"

SNMP port
Sets the UDP port on which the device is responding to SNMP requests.
Default: "161"

sysContact
Defines the contact of the person responsible for managing this device via SNMP.
Default: "root@this.ip.com"

sysName
Defines the SNMP system name of this device.
Default: "Barix SIP OPUS device"

sysLocation
Defines the SNMP system location of this device.
Default: "Please update me!"

Read-Only Community
Set here the name of the community, that will have read access to the SNMP values, reported by this device.
Note: When the Read-Only Community name is not explicitly set (empty), the name "public" is assumed.
Default: "public"

Read-Write Community
Set here the name of the community, that will have RW access to the SNMP values, reported by this device.
Note: When the Read-Write Community name is not explicitly set (empty), the SNMP SET function is disabled (no write access).
Default: "private"

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I/O CONTROL SETTINGS

Enable I/O Serial Tunnel
Enables the TCP-to-serial tunnel.
Default: "Disable"


TCP Tunnel Mode

Adjust the tunnel mode:
Server: In this mode the SIP client is waiting for a telnet connection on the specified port. Once established, data can be transmitted between the configured COM port, and the host that has established the TCP connection.
Client: In this mode if the SIP client receives any data on the configured COM port, it tries to establish a TCP connection to the specified address and port. Once TCP connection established, data can be transferred between the serial port and the TCP remote host.

TCP Control IP Address
Configure here the TCP address and port for the active TCP Serial Gateway.
Default: "127.0.0.1"

TCP Control Port
Sets the TCP control port on which the TCP Server shall listen to, or the TCP Client shall try to connect to.

Default: "12301"

Port Name
Configure here the device name of the serial port in use.
Default: "/dev/ttyS1"

Baud Rate
Select the serial transmission speed ("9600" to "115200" Baud).
Default: "9600"

Data Bits
Select "7" or "8" data bits.
Default: "8"

Parity
Select "None", "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: "Off"
RS232: "Software flow control(XON/XOFF)" or "Hardware flow control (RTS/CTS)"
RS485: "RS485 direction control"
NOTE: On MA400 devices (SIP OPUS Codec) this setting is ignored, since the RTS signal is used as "door relay".
Default: "none"


SECURITY SETTINGS

Reboot Function
Enable or disable the "Reboot" function with the RESET button and from the WEB UI REBOOT tab.
Default: "enabled"

Reset Factory Defaults
Enable or disable the "Reset Factory Defaults" feature from both web UI "DEFAULTS" tab and device front panel RESET button. In order to revert all settings to factory defaults click on the "Defaults" web UI page, or keep the Reset button pressed for more than 10 seconds.
Default: "enabled"

Update Function
Enable or disable the WEB Update function of the device.
Default: "enabled"

Web UI Password
This group of fields is visible as long as no password is set.
Enter the admin password (up to 16 characters) both in the "Password" and "Confirm Password" fields, then click the "Submit" 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. Use the default admin user name and the password that you have just set in order to access the web UI.

Change Password
This group of fields is visible as long as a password is set.
To allow free access (clearing the password) enter the old password and leave the fields "New Password" and "Confirm Password" empty. After the restart you will not be asked for user name and password anymore.