Help with SIP Trunk and Codec Negotiation

Hello, I am using Asterisk 15.5.0
When I make a call from a soft-phone I have, I get this error on my Asterisk CLI:

 -- Called SIP/18004444444@trunk-1
    -- SIP/trunk-1-00000043 is making progress passing it to SIP/103-00000042
    -- SIP/trunk-1-00000043 answered SIP/103-00000042
[Aug 19 12:20:48] WARNING[22212][C-00000026]: channel.c:6560 ast_channel_make_compatible_helper: No path to translate from SIP/trunk-1-00000043 to SIP/103-00000042
[Aug 19 12:20:48] WARNING[22212][C-00000026]: app_dial.c:3178 dial_exec_full: Had to drop call because I couldn't make SIP/103-00000042 compatible with SIP/trunk-1-00000043
  == Spawn extension (group-1, 18004444444, 6) exited non-zero on 'SIP/103-00000042'
vps194854*CLI>

I have configured my softphone to only use ulaw and alaw all other codecs are disabled.
I yesterday, I bought 1 G.729 license, let me show you:

vps194854*CLI> g729 show licenses
0/0 encoders/decoders of 1 licensed channels are currently in use

In the sip.conf I have this:

[103]
type=friend
accountcode=xxxxxxxxxxx
context=group-1
secret=xxxxxxxx
host=dynamic
disallow=all
allow=ulaw
allow=alaw

[trunk-1]
type=friend
defaultuser=xxxxxxxxx
secret=xxxxxxxxxxx
host=sip.xxxxxxxxxxxxx.com
disallow=all
allow=g729

Why am I getting this error?

sip set debug on on and check the codec negotiation