The FONESTAR ZSM system is now a SIP client
A new and extremely appealing feature in high demand in the ZSM system:
The ZSM-1050 and ZSM-1000 can now act as SIP clients.
“Session Initiation Protocol. This is a protocol developed by MMUSIC of the IETF with the goal of becoming the standard for the initiation, modification and completion of interactive user sessions involving multimedia elements such as video, voice, instant messaging, online games and virtual reality.”
The SIP protocol is widely used as a digital telephone switchboard.
From now on, ZSM devices can act as phone extensions. When you call this extension, a voice announcement will be made via public address.
This makes the devices very useful as they combine independent functionalities such as background music, and messages and events along with the possibility of delivering voice announcements via an ever expanding standard.
- A SIP server is required.
- To be able to configure the equipment the ZSM-Go program must be of version 3.0 or higher
- Both the ZSM-1050 and ZSM-1000 devices must be of version 2.0 or higher
- Program and equipment update files are available for you to download on the Fonestar website under the tab “Software” for products
- Please note that for the systems that have already been installed and for which the update will be applied:
- This is a large update that takes 20 minutes to complete.
This is done using the ZSM-Go program.
Once connected to a unit, select the “SIP” tab within the unit configuration:
To the left you can see the equipment. A “SIP” icon has been added which has 3 possible state:
- Orange: Trying to register on the server.
- Green: Successfully registered on the server (you can now receive calls).
- Blue phone icon: Call in progress.
On the right you can see the “SIP” tab where the configuration parameters have been added:
- Activated: The checkbox is activated.
- Server parameters:
- IP: Server IP. It can be in the same local network (LAN) or it could be an external internet address.
- Port: Communications port of the server.
- User: User associated to the extension that is assigned in the server.
- Password: Password associated to the extension assigned within the server.
- Audio port range: Ports that are used to send/receive audio.
- Device configuration:
- Active zones during SIP calls (ZSM-1050 only): Indicates in which zones the calls will come through. In the case of the ZSM-1000 they will come through the single audio output.
- Relays active during SIP calls (ZSM-1050 only): Indicates which relays are activated during the call.
- Volume: The volume at which the call will be played back.
The priorities of the different functions, from highest to lowest, are as follows:
- SIP call.
- Events (only ZSM-1050).
- Programmed and instant messages.
- Background music.