Sip Trunk setting issue

I defined my first Sip Trunk and assigned it to an predefined outbound route, but when I dial a number that match the route dial plan from an extension, I see that sip invite to asterisk and then 100 Trying and Next 183 Session Progress and after RTP for busy lines form asterisk, server received Cancel from Ip Phone and after ACK from extension Call ends. There is no Sip Invite has been send from asterisk to other side of Sip Trunk. only after the calls ends, I see a OPTIONS sip: to opsite side and it answers with 200 OK.
It seems , there is a problem with Sip Trunk setting. I think that asterisk can find outbound route and it’s assigned sip trunk, but call does not properly proceed.
Can anyone help me, Why?
Thanks in advanced

Can you post your asterisk CLI output during a test call? Can you verify that the number you’re trying to dial is getting to asterisk as expected, and not being interfered with an incorrect dialplan/digitmap on the phone?

Asterisk never attempted to access the trunk.

After a failing test call, go to Reports->Asterisk Logfiles and view the dialplan.

If you heard the equivalent of “Your call cannot be completed as dialed”, this means that the number did not match any Outbound Route, either as @wmoon said, or because the route is not set correctly.

If you heard “All circuits are busy now”, this means that the trunk is not registered (if it’s a trunk requiring registration), qualify failed (if you have qualify enabled for the trunk), or you have a low level networking issue (DNS lookup failed, no route to host, etc.)

Thank you @wmoon & @Stewart1 for attentions. Do you mean the playing file for “all-circuit-busy-now.ulaw”? Yes, it’s there . Do you mean that sip trunk does not registered? The sip trunk registeration in both sides are disable and I can see OPTION packet and it’s 200 OK response in the network monitor with WireShark.

Here is CLI report for a test call:

Thank you guys. With your trigger I found it. There was a carbage route!

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