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In the tab, you configure settings for apartment, device, calls, and SIP calls.
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Apartment Settings
For correct panel functioning, you must enter information about its logical address (for more information
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, see Logical addresses formation and setting):
panel operation mode (choose Unit mode for correct work)
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building number
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unit number
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device number
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If you have more than 1 entrance panel with the same logical address, you need to specify their device numbers, starting from 1 to 9. |
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Note: On a multi-button entrance panel, each button corresponds to an apartment from 1 to N (depending on the model). To make a call between the panel and the indoor monitor by pressing the button of the required apartment, the panel and the monitor must have the same building and unit number in apartment settings. Also for monitors, you must indicate corresponding information about the floor, apartment, and device number. According to the panel settings from the photo above, when you press the button of apartment No 1, located on the 1st floor of the 1st unit, a monitor with 0001-01-00-01 logical address (number) will be called. Thus, the same logical address must be set in the monitor settings: building - 1, unit - 1, floor - 0, apartment - 1. If there are no such settings, you must |
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make forwarding queues |
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for apartments with IP addresses or SIP numbers of monitors. |
SIP settings
These settings are required for the correct work of calls via SIP protocol. If you use
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the BAS-IP Link server we recommend configuring SIP settings remotely using the Link (via MQTT protocol) and then sending them to the device.
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For more info on how to get SIP
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numbers and configure a panel for SIP calls if you use a BAS-IP SIP server
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, refer to the Step-by-step guide.
To configure SIP calls for the panel, you must enable device SIP registration and enter the following parameters:
SIP server proxy that can be represented by both an IP address and a domain name
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Data format: Before the proxy address, you must enter "sip:", e.g. sip:gb.sip.bas-ip.com. A full list of BAS-IP servers for each country is available here. If you use a third-party SIP server, you can also indicate a non-standard port in the format: sip:gb.sip.bas-ip.com:1506 where gb.sip.bas-ip.com is a SIP server proxy, 1506 is a non-standard port. |
SIP server address that can be represented by both an IP address and a domain name
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, e.g. gb.sip.bas-ip.com
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Data format: If you use a third-party SIP server, you can also indicate a non-standard port in the format: sip:gb.sip.bas-ip.com:1506 where gb.sip.bas-ip.com is a SIP server, 15061 is a non-standard port. |
server STUN IP address. For example, stun.l.google.com
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port of the STUN server
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Note:
19302 port is used for the Google STUN server.
user SIP number (up to 20 characters)
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SIP user ID (optional)
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password for SIP number (up to 20 characters)
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transport type: UDP/TCP/
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TLS
Note:
UDP is good for speed and efficiency.
TCP is great for reliable IP-packet delivery.
TLS is best for encryption, authentication, data integrity, and secure SIP trunking.
If the connection with the SIP server is lost, it will be automatically renewed. You can set the re-registration interval (30-900 seconds) between attempts to renew the connection.
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Call settings
At this part, you can perform the following actions:
enable/disable a call to the specified number when pressing the concierge button
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If the feature is disabled, the concierge is called via an internal protocol. If the system has concierge monitors, the call will go to the main monitor. If it does not answer, the call will be transferred to the other monitors in the system (if they are).
indicate the direction (number) to which the call will be made when pressing the concierge button
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Forwarding when pressing the concierge button: You can also set up a call to a specified number/s (up to 8) when you press the concierge button using the Forwarding feature. In the Apartment number field, you must enter 1000Х, where X is the concierge monitor device number. So, for the 1st concierge monitor apartment number will be 10001. Also, you must enter the forward number in one of the formats: sip:SIP number@address of the SIP server (if you need calls to be done via SIP protocol) or sip:any number@device IP address (if calls must be done via P2P protocol). |
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configure call max time
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(10-120 sec
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after which the panel automatically ends an outgoing call if there is no answer).
configure talk max time
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(10-300 sec
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after which the panel automatically ends the outgoing conversation).
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If the feature is disabled, the concierge is called via an internal protocol. If the system has concierge monitors, the call will go to the main monitor. If it does not answer, the call will be transferred to the other monitors in the system (if they are).
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Talk time for an incoming call is limited to 2 min.
Device settings
In this section you can:
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Talk time for an incoming call is limited to 2 min. |
enable/disable the end of outgoing calls when (connected to the panel) the exit button is triggered (e.g., by the concierge).
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Device settings
In this section, you can configure the following parameters:
preferred video quality (resolution) (640х480/1280х720/1920x1080 (optional))
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preferred RTP data profile
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the volume level of a panel speaker
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RTSP username (login to get access to a panel RTSP stream)
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RTSP password (password to get access to a panel RTSP stream)
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enable/disable audio for RTSP stream
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enable/disable a proximity sensor for the panel automatic turning on the keyboard backlight and face recognition feature when motion is detected at a distance of 50 cm
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select the appropriate proximity sensor mode:
all-time mode: the sensor is active all the time and triggered when motion is detected at a distance of 50 cm
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adaptive mode (appropriate when the panel is installed in front of a wall or an object): the sensor gets used to a constant object in front of it and does not react to it, but only triggers when there is movement
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enable/disable connected to the panel temperature sensor. During an outgoing call, the panel will take the temperature of the person in front of the panel and display it on the called person internal monitor during the call/talk.
