Unable to dial out through remote sip trunk (fxo card)

[file/] I’m new to asterisk, but have read Asterisk forum and my VoIP equip manuals.
I have a Draytek Vigor 3300V+ router with a 4-port FXO card installed on IP 10.10.10.1
My asterisk server is 10.10.10.4

All my extensions are registered and can dial one another without a problem. However, I can not figure out how to get the FXO card in my router to connect my PSTN lines via SIP to my asterisk box.

The router manual is here:
draytek.com/user/SupportAppnotes.php?Id=83
The notes on Asterisk / FXO functionality do not make sense to me and need help from you guys!! I have searched everywhere for help but am lost. I even tried this manual config mentioned here: draytek.com/user/SupportAppn … .php?ID=85

When I configure the FXO card I give it the config shown in the attachments.
SIP Protocol Configured like this:

SIP Account for FXO / PSTN card config like this:

This shows my FXO ports active and ready

This is my FXO port1 config

In asterisk my set up my sip account for trunk 1 like this:

host=10.10.10.1
username=1001
secret=1001
type=friend
qualify=no
insecure=very
dtmfmode=rfc2833
disallow=all
context=from-pstn
allow=ulaw&g729&g723.1
trustrpid=no
sendrpid=no
canreinvite=no

and my register string:
1001:1001@10.10.10.1/1001

I set up this via the “Custom SIP Trunk” in freepbx

Help!!!

EDIT:
When I dial out: “All circuits are busy now”

Which version of Asterisk are you using? Although it only affects inbound calls, insecure=very will be ignored by current versions.

If the all circuits are busy is the one that as four numbers, three in parentheses, you need to look for the real error, which will be further up the log.

Why are you registering (both sides) when all the addresses are static?

For a SIP problems you need to include the SIP and SDP that is actually exchanged, you also need to include the actual error message, and include the complete message, not just the fixed text.