Having trouble with testing asterisk-18 pjsip.conf

Asterisk Asterisk SIP
1

Hi All,
asterisk-18 for installed into a system, few years ago. Now, nobody worked on that project is available, I could not see sip.conf in that system and seeing pipsip.cong, extensions.conf, http.conf, asterisk.conf. as per my understanding, we define sip peers in sip.conf and mention that in extensions.conf like
sip.conf
[7001] ;peer/channel
username
secret
hostname

extensions.conf
exten => 7001
but i am not finding that association in my case, it looks like below, where i am not able to understand where they configured XYZ and ABC, could someone shed light on this and i am in an environment where i cant download any softphones to test these configuration, i would like to know the existing configuration is working or not, could you suggest the way to test these config plz?

pjsip.conf
[XYZ]
extensions.conf
+123456789
same => n,Dial(PJSIP/12345678@ABC)

2

If this represents a normal telephone, there should be no “username”, and hostname should have the value dynamic. username is a deprecated option name and it does not do what people think it does, which is why its name was changed. More generally, if you are looking at someone else’s chan_sip configuration, it is likely to have multiple errors; you need to understand what the configuration is doing.

This looks like trying to dial through an internet service provider. For a local telephone, the dialstring format is the same as for chan_sip, except with PJSIP/ instead of SIP/.

Typical chan_pjsip configurations for simple cases are given in res_pjsip Configuration Examples - Asterisk Documentation

Some of the more commonly used chan_pjsip dial string formats are given in Dialing PJSIP Channels - Asterisk Documentation