Suitable for any traditional 4-5 wire legacy doorphone with the possibility of connecting it also to the user's internal intercom station
The AA-15IP device allows to interface any traditional doorphone (4 or 5 wires) to an IP-PBX in order to manage the doorphone from any internal extension. When a visitor presses the call button of the existing doorphone, AA-15 generate a SIP call to a programmed extension number by putting the visitor in communication with the operator who answered, who has the possibility to operate, with a code from the phone, the relay for opening the entrance gate. A second relay can be activated, from the internal telephone as well, for other services.
In the presence of the IP-PBX, calls from the intercom can be routed to an external telephone number, including mobile numbers.
The device has an external contact input that can be used as an alarm dialer: when closed, AA-15IP calls a programmed number and alerts the event with a customizable voice message.
If there is no IP-PBX, the basic functions (communication and gate opening relay activation) are obtained by connecting a normal IP-SIP phone and programming AA-15IP in P2P (Peer-to-Peer) mode.
It is possible to connect AA-15IP both to the central unit of the doorphone system or directly to the internal station (wall-mounted intercom) of the single user.
Tema AA-15IP integrates a PoEpower supply and can therefore be powered on the same cat5/6 LAN cable if coming from a PoE switch. Alternatively, an input for external 230Vac power supply is available (Optional).
The dimensions 76.5x62xH32.5mm (connectors excluded) are extremely small and the system can be fixed to the wall or on a DIN bar.
Main features
- Can be connected with all 4-5 wires doorphones models, DIN rail mounting
- 1 power open-door relay and 1 auxiliary relay
- Up to 2 configurable extension numbers (in case of no answer)
- 3 alternative programmable numbers according to day/night/interval times
- Relay: configurable opening and closing contact
- Activity display LED
- Easy programming via Web browser
- Compatible with the most popular IP-PBX brands
Technical Specifications
- LAN, protocols: TCP/IP Network 100BaseTx, SIP 2.0
- Grade of protection: IP20
- Power supply: PoE, Injector PoE o external power supply
- PoE: 802.3af class0 12,95W
- External power supply (Opz.): 230Vac/12-Vdc-
- Consumption: 500mA
- Technology: IMX25400 MHz processor
- Storage: b 256MB Ram, Micro-SD Flash 8GB
- Capacity of the 3 internal relays: max 30Vdc -1,5A
- Housing material: ABS
- Operating temperature :from -20° to+55°C
- Storage temperature: from-20° to+65°C
- humidity: up to 90%
- Installation: table, wall-mounting, standard DIN bar
- Audio Amplifier: 2W"D" Class
- Connection: SIP Server (IPPBX) orP2P (Peer To Peer)
- Audio communication: Bidirectional
- Inputs fromexternal control: 1 (Alarm)
- Command type: voltage-free contact, transistor O.C.
- Visual signals: Call active Led, relaysactivated
- Programming: web interface and password
- Adjustments: receiving volume adjustment
- Mounting: wall-mounting with accessory supplied, DIN bar
- Housing: ABS
- Dimensions: 76,5x62x32,5mm
- Warranty : 2 years
- Certification: CE, ROHS
When the visitor presses the button, AA-15IP generates a SIP call to aninternal number, putting the visitor in communication with the operator, who has the possibility to operate the gate opening relay with a code from the phone. In any case, the inter-nal doorphone station remains operational since AA-15IP works in parallel.
The contacts of Relay 2 can also be brought to the internal station to allow the opening of a second gate or to switch on anylights.
Both relays can be activated not only following the call from the doorphone but also by calling AA-15IP to the assigned num-ber/IP address, wait for the answer and activate them with codes from the telephone keypad. In the presence of the IP-PBX, calls from the doorphone can be routed to external phone numbers, including mobile.
Note:Alternatively, the SIP relay function can be used with Tema AD611 or AD612 by using the second account (ringer group) and programming it in Relay1 or Relay2 (or both). The switching contacts close when a SIP call is received (see the following screenshot).
