
ARA-1 Operations Manual
4-2
INTEROPERABILITY NOW
4.3.2
Basic Operation
The ARA-1, once fully and properly configured, appears as a SIP endpoint and will respond to
SIP invites. If it has been programmed to register with a SIP proxy, it will attempt to do so. Once
a connection is established, the LINK ACTIVE indicator on the front panel will light. When the
unit receives audio via the SIP connection, it will key the transmitter via the PTT line, and the
audio will be transmitted over the radio link. When the associated radio is unsquelched and causes
the ARA to detect active COR, the CHANNEL ACTIVE indicator on the front panel will light,
and the received audio will be sent over the SIP network.
NOTE:
What causes the ARA to detect active COR depends on the method used
to detect that the radio is receiving a valid signal: VOX or a hardwired COR input
from the radio. This is controlled by the
Radio COR Type
setting. See Section
4.3.3
Outgoing Call Initiation
There are three methods to initiate a call from the radio side rather than the network side:
•
Call Initiation via DTMF
•
Call Initiation via COR Cadence
•
Web browser
DTMF call initiation involves the use of a radio’s DTMF keypad to transmit a preset sequence
of digits that correspond to a SIP extension or the IP address of a SIP end user. The DTMF
sequences and corresponding SIP destinations are pre-programmed per the procedure explained
in Section 0. Another DTMF sequence terminates the call.
COR Cadence call initiation involves the use of preset squelch break sequences; that is, a remote
radio user triggers the radio’s PTT input a preset number of times at a defined rate. The radio
cabled to the ARA-1 detects these transmit bursts and signals the ARA-1, which initiates a call
to the preset SIP destination that corresponds with the detected cadence. The COR Cadences and
SIP destinations are pre-programmed per the procedure explained in Section 3.4.2. Another COR
Cadence terminates the call.
Calls can also be initiated by browsing to the ARA-1’s IP address, selecting the
Call Management
page, entering the SIP destination (SIP PBX extension number or the end-device IP address) and
pressing
Connect
. See Figure 4-1.