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Depending on the version of Asterisk that you are using, You may have two channels drivers that you could use in order to create a peer that you could use to place and receive calls, if you are looking for how to configure asterisk with chan_sip we have another KB article that talks about the configuration.
Asterisk with chan_pjsip
To configure Asterisk's PJSIP-based SIP channel driver, included with Asterisk versions 12, 13 and newer, to work with Digium's SIP Trunking service, you should configure 6 objects:
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Here, in the digium-siptrunk-identify object, we've declared that we'd like to match inbound calls from sip.digiumcloud.net (the Digium SIP Trunking server) to the digium-siptrunk endpoint (the one we've created here).
Asterisk Dialplan Configuration
Asterisk uses a configuration of numbers and patterns matched with specific actions, called dialplan, and configured in extensions.conf, typically located in /etc/asterisk in order to determine what to do with incoming calls. In order to complete your setup with Digium's SIP Trunking service, you will need to properly configure your Asterisk dialplan. To learn more about Asterisk's dialplan, please see the Dialplan wiki page, as well as its children, available on the Asterisk wiki.
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