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Table of Contents

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 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 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);
  • sync code between monitors. 


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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. 


SIP 

The SIP settings are required for the monitor to work via the SIP protocol. 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. 

  • user SIP number;

  • password for the SIP number;

  • RTP data profile;

You can enable the auto re-registration to renew the lost connection with the SIP server and enter the re-registration interval (from 5-600 seconds). The feature is recommended to use in networks with an unstable Internet connection. If necessary, you can re-register the device manually by clicking the appropriate button.

DTMF settings 

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:

  • enable/disable opening the lock before answering a call;
  • 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.

Advanced Intercom settings 

n 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;
  • speaker volume adjustment during a call (talk volume);
  • setting the conversation maximum time.

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:

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

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: 

  • 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:

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

 

  • 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 period of 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.

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.

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 current supply;
  • enable/disable current supply during a call and conversation.