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How Calls Are Processed
Page 67
CHAPTER 7
How Calls Are Processed
In the following discussion, SIP and PSTN endpoints/interfaces are interchangeable, and they may be used on no,
one, or two sides of the call.
Diva SIPcontrol uses an endpoint-based approach to process calls, which means that every PSTN interface and
every configured SIP peer is considered as a single endpoint. The endpoint holds Diva SIPcontrol settings for
the respective PSTN interface or SIP peer. Each call originates at a specific endpoint (on the SIP side after
assigning the SIP call request to one of the configured peers) and needs a route to find its designated endpoint
(the destination). Thus, the most simple configuration needs two endpoints of any type and one route, as shown
in red in the graphic below.
This graphic shows that an endpoint is only a virtual object of a real device. The endpoint holds the settings for
the corresponding device. For example, if a call should be routed from SIP Device 3 to PSTN Device 2 as marked
red in the graphic, then the:
Settings of SIP Device 3 need to be configured as a SIP peer endpoint in the SIP Peers section.
Settings PSTN Device 2 needs to be configured as a PSTN endpoint in the PSTN Interfaces section.
Condition "called address is 3456" needs to be configured in the Routing section to route the call to the
correct device.
For example, if you have a SIP or PSTN Device 4 with no endpoints configured in Diva SIPcontrol, you cannot
establish a call, because Diva SIPcontrol will not know the settings of the device.
The PSTN endpoint is found via its controller number. On the SIP side, multiple SIP peers can connect via the
same network interface. Therefore, the assignment is more complex:
1. The host/domain name and port number of the received "FROM" header is compared against the SIP peer
settings.
2. If no host matches, the same address is compared against the "Domain" parameters of the SIP peers.
3. If no match is found, Diva SIPcontrol looks for a SIP peer with the Default SIP Peer option enabled.
4. If the call cannot be assigned, regardless of whether the call originated in the PSTN or SIP network, the call
is rejected.
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