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In the tab, you have access to communication, logical address, SIP, and DTMF settings. Some of these settings can be also set via the monitor settings.
Address Settings
For correct device functioning, you must enter information about its logical address. The logical address is used to automatically synchronize and search for other devices in the network and is also necessary to provide a connection between them (for more information, see Logical addresses formation and setting). If you use the BAS-IP Link server we recommend configuring address settings remotely using the Link (via MQTT protocol) and then sending them to a device.
You must enter the following data about the device location:
building No. (from 0001 to 999);
unit No. (from 00 to 99);
floor No. (from 00 to 98);
room No. (from 01 to 99);
device number (from 0 to 8);
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If you have only one number enter 0 for the device number. If you have more then 1 the numbering must start with 0 and end with 8. |
sync code between monitors.
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SIP
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The SIP settings are required for the monitor to work via the SIP protocol. If you use the BAS-IP Link server we recommend configuring SIP settings remotely using the Link (via MQTT protocol) and then sending them to the device.
For more info on how to get SIP numbers and configure a panel for SIP calls if you use a BAS-IP SIP server, refer to the Step-by-step guide.
To configure SIP calls for the monitor, you must enable device SIP registration and enter the following parameters:
SIP server proxy (where the device will send registration requests) 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:us.sip.bas-ip.com. A full list of BAS-IP servers for each country is available here. |
SIP server address (used to send a domain in an authentication request) that can be represented by both an IP address and a domain name, e.g. us.sip.bas-ip.com.
server STUN IP address. For example, stun.l.google.com;
port of the STUN server;
Note:
19302 port is used for the Google STUN server.
transport type: UDP/TCP/TLS;
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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. |
user SIP number;
password for the SIP number;
RTP data profile;
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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DTMF settings
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In the menu, you can configure the DTMF code, entering which the lock will be opened. Code length is up to 4 characters.
In the section, you can perform the following actions:
enable/disable opening the lock before answering a call (works via internal protocol);
set DTMF code for lock #1.
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By default, all entrance panels are configured to receive the # DTMF code to unlock the first lock. |
If you have 2 locks (e.g. with the help of the SH-42 module), you also can:
enable/disable second lock functioning;
set DTMF code for lock #2.
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By default, all entrance panels are configured to receive the 0 DTMF code to unlock the first lock. |
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Advanced Intercom settings
In this tab, you have access to the configuration of the following parameters:
enabling/disabling all incoming calls autoanswer;
enabling/disabling auto hung up after pressing the Open button during a call;
setting the delay time after pressing the Open button before opening the lock;
speaker volume adjustment during a conversation (talk volume);
general volume adjustment;
setting the conversation maximum time.
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Call forwarding
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With this feature, you can connect up to 4 devices that will receive calls simultaneously with a call to the internal monitor.
For feature functioning, you must click Add and enter the number/s in the URL field. When all data are entered, submit settings.
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Number format for calls via P2P:
Number format for calls via SIP:
For correct functioning, the SIP numbers must be registered with the same SIP server. |
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Concierge call
Here you can enable/disable the feature of a quick call via SIP or P2P to an indicated number or concierge.
2 modes are available:
call to specified number: when the concierge button is pressed, a call will be made to the number you specified (via the P2P or SIP protocol). It can be the number of concierge or any other device. The button for the call is in the navigation bar of the monitor main screen.
To configure this mode, you must select it in the corresponding field, enter the number in P2P or SIP format, and submit settings.
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Number format for calls via P2P:
Number format for calls via SIP:
For correct functioning, the SIP numbers must be registered with the same SIP server. |
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multiple-concierge call: call all concierge monitors (AM-02) that are in the same network via the internal protocol. The call is made in turn to each monitor after a specified time (by default in 5 sec), which you must indicate in the Call interval for each concierge field. The feature will work after saving the settings.
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Incoming call indication
Here you can enable/disable the supply of the monitor alarm output with +12 Volts after receiving an incoming call. For example, a lamp can be connected to the monitor alarm output and during an incoming call, the lamp will light up. It will be as additional incoming call signaling, e.g., for people who are hard of hearing.
You can configure the following settings:
enable/disable an incoming call indication;
enable/disable the current supply turning off after answering a call;
set the duration of the current supply;
enable/disable current supply during a call and conversation.
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Multicast paging
Multicast pagingdistributes datato multiple recipients.Using multicast, a device (monitor) can send data (e.g., notifications or calls) to a provided multicast IP address, which is then distributed to other devices subscribed to listening to the specified IP address.
For the feature operation, you must enable the multicast paging and provide an IP address and port that the monitor must listen to get notifications.
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