Intercom

1 Address Settings
2 SIP Settings
3 DTMF
4 Advanced Intercom settings
5 Call forwarding
6 Concierge call

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:

building No. (from 0001 to 9999);
unit No. (from 00 to 99);
floor No. (from 00 to 98);
apartment No. (from 01 to 99);
device number (from 0 to 8);

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

sync code between monitors.

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 on 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 that can be represented by both an IP address and a domain name.

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 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;

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.

maximum talk timeout;
user SIP number;
SIP user ID (optional);
password for SIP number.

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

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).

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.

By default, all entrance panels are configured to receive the 0 DTMF code to unlock the first lock.

Advanced Intercom settings

In this section, you can enable/disable the following features:

turning on/off the monitor camera (camera is not available for monitors with the letter L in the name);
enabling/disabling the feature of automatic call acceptance by the monitor in the absence of the answer. If there is no response within 30 seconds, the monitor will start recording a message from the visitor (answerphone feature);
enabling/disabling of deaf mode (the monitors alarm output will be supplied with +12 Volts after receiving an incoming call);
enabling/disabling all incoming calls autoanswer;
enabling/disabling auto hung up in 3 seconds after pressing the Open button during a call;
enabling/disabling the lock protection from accidental opening. A user password will be required to open the lock during a call;

Answerphone feature is 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.

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.

Concierge call

Here you can enable/disable the feature of 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.