To start using SipPhone plugin in TRASSIR, the below described settings should be configured. After that, TRASSIR server operator will be able to call Asterisk subscribers, receive calls and send service instructions.
Important
First of all, for each TRASSIR server, an account on Asterisk server should be created. TRASSIR server operator will use it to receive and make calls.
Set up Connection parameters:
- Asterisk server - Asterisk server IP-address or DNS-name.
- Asterisk port - Asterisk server network port (by default:
5060). - User and Password - account name (phone number) and password on Asterisk server.
- Activate DND and Deactivate DND are commands sent to server to activate and deactivate DND ("Do Not Disturb") mode
- Key - a command to open the door sent to home entry system device from operator's interface.
The status of connection to dial office IP is displayed in the Status field. In case all parameters are specified correctly, Connected line will appear. Otherwise you'll see error message.
In case you would like the current server operator to have access to the calls history, phone talk records and set associations of channels, select in Master TRASSIR field name of server on which connection to AMI server is set.
Tip
Selecting a server as Master Trassir will be possible only after the connection to it. See detailed description of connection to server procedure in the section Connecting to a new server.
