Hide caller's id

Hi, I have a trunk sip and when I try to call through the trunk it fails because the asterisk sends the extension that the call is made and in the other asterisk does not exist, I really need to send the id with which the account was registered.
I have read that the variable extend can only be modified with Goto, can someone tell me what the syntax is like?
Thanks.
Regards.

Not quit clear to me what you are asking…
But a quick way to set a callerid for outgoing calls is : exten => _X.,n,Set(CALLERID(num)=0ZXXXXXXXXX)

0ZXXXXXXXX the number assigned to your sip trunk by your provider.

${EXTEN} is not send to the peer, except as a result of soemthing explicit in the dialplan, e.g. Dial(SIP/${EXTEN}@peer).

Thanks for the reply.
The provider from another country gave me a sip account to configure in my asterisk and calls to that country should use that account.
Set up the account in sip.conf and register correctly.
In the extension.conf configure that the callers use the configured trunk and change the callerid:

TRUNK_INTERNATIONAL_F=SIP/TRK-FRANCE
exten => _90033X.,1,Set(CALLERID(name)=Trunk_1)
exten => _90033X.,n,Dial({GLOBAL(TRUNK_INTERNATIONAL_F)}/{EXTEN},120,TtrwW)

The account uses a name (Trunk_1) and not a number

I make the call and show this message:

chan_sip.c:20257 handle_response_invite: Failed to authenticate on INVITE to '"Prueba" <sip:1000@xxx.xxxxx.com>;tag=as5a47807c'
    -- SIP/TRK-TRUNK_INTERNATIONAL_F-00007025 is circuit-busytype or paste code here

My asterisk sends the internal callerid and in the remote asterisk does not exist.

Any suggestions?

Thank you.

Set fromuser in sip.conf.

Thanks for the help, get my asterisk to send the callerid they gave me. I can’t call but the message changed a little

NOTICE[8283]: chan_sip.c:20257 handle_response_invite: Failed to authenticate on INVITE to '"Prueba" <sip:Trunk_1@xxx.xxxxx.com>;tag=as3f80cafa'
    -- SIP/TRK-TRUNK_INTERNATIONAL_F-0000793b is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Auto fallthrough, channel 'SIP/1000-0000793a' **status is 'CONGESTION'**
  == End MixMonitor Recording SIP/1000-0000793a

Is a remote asterisk problem?
Sip account is registered correctly.
Another suggestion?

Thanks