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In the tab, you have access to communication, logical address, SIP, and DTMF settings. These settings can be set both in this tab and 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 (more information about logical address formation find here). 

You must enter the following data about the device location:

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If you have only one number enter 0 for the device number. If you have more then the numbering must start with 0 and end with 8. 

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SIP Settings

These settings are required for the correct work of calls via 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. 

Step-by-step guide how to get SIP nubmers and configure a monitor for SIP calls if you use BAS-IP SIP server you can find here.

To configure SIP calls for the panel, 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;

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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. 

• re-registration interval to renew the lost connection with the SIP server;  
• maximum talk timeout;
• user SIP number;
• SIP user ID (optional);
• password for the SIP number.

You can enable/disable SIP early media. If connection with SIP server is lost, it will be automatically renew. You can set the re-registration interval (from 30-900 seconds) between attempts to renew the connection.

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DTMF 

In the menu, you can change the DTMF code, entering which the lock will be opened. Code length is up to 4 characters. 

In the section, you can configure:

• lock #1 name that is displayed during a call;
• enabling/disabling opening the lock before answering a call when pressing the unlock button (works via internal protocol);
• set DTMF code for lock #1 (if default value is disabled). 

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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;
• enter lock #2 name that is displayed during a call;
•  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 section, you can enable/disable the following features:

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Answerphone and Automatic call recording features are not available if an SD card is not installed, or if it is not the main monitor.

Call forwarding 

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: 

• sip:1@192.168.1.65, where 1 is the desired number to be displayed for the callee, 192.168.1.65 is the IP address of the callee SIP client (if you use a softphone, the IP address of a device where the softphone is installed);

Number format for calls via SIP:

• sip:5588@us.sip.bas-ip.com, where 5588 is the callee SIP number, us.sip.bas-ip.com is the address of the SIP server, which can be either the IP address or domain name.

For correct functioning, the SIP numbers must be registered with the same SIP server.

If necessary, you can delete one or all numbers.

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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, 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: 

• sip:1@192.168.1.65, where 1 is the desired number to be displayed for the callee, 192.168.1.65 is the IP address of the callee SIP client (if you use a softphone, the IP address of a device where the softphone is installed);

Number format for calls via SIP:

• sip:5588@us.sip.bas-ip.com, where 5588 is the callee SIP number, us.sip.bas-ip.com is the address of the SIP server, which can be either the IP address or domain name.

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, which you must indicate in the Call interval for each concierge field. The feature will work after saving the settings.

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• when the function is disabled, the call will be made to the first found concierge monitor (AM-02) in the network.

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