We don’t support GUIs here, so please provided configurations as text files.
bindport, on chan_sip, cannot be used for an individual peer; it must be in the general section.
Why do you believe you need to set nat=force_rport, comedia, when you haven’t mentioned any NAT, and your configuration doesn’t have the external address information needed to support Astierisk behind NAT? (You’ve used a deprecated way of specifying these options.)
Do you really have a SIP proxy at a.a.a.b? Or is this just an IP gateway?
You don’t appear to have any registration information for chan_sip, but, if I’ve deciphered the GUI correctly, you do for pjsip.
You have an outbound proxy in the chan_sip configuration, but not the pjsip one.
You don’t appear to have any NAT workarounds in your pjsip configuration.
“the number is not answering” but the call didn’t go through.
This message is not a standard behavour of Asterisk. I presume it is generated by code provided as part of your GUI; you need to get support from the GUI community, e.g. https://community.freepbx.org, for FreePBX.
For the avoidance of doubt, you should not configure pjsip and sip at the same time, unless they are bound to different port numbers or interfaces.