Unable to create channel of type 'SIP

[Joss2103]

Hi, sorry for the inconvenience, but is it possible that you can help me with this error please?

app_dial.c:2455 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

This is the complete trace

== Using SIP RTP CoS mark 5
-- Executing [412311143483@default:1] Macro("SIP/10001-00000005", "salida-ift,SIP/BLASPavel_TRUNK,2311143483") in new stack
-- Executing [s@macro-salida-ift:1] AGI("SIP/10001-00000005", "agi://127.0.0.1:4577/call_log") in new stack
-- <SIP/10001-00000005>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [s@macro-salida-ift:2] AGI("SIP/10001-00000005", "ift.php,2311143483") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/ift.php
-- <SIP/10001-00000005>AGI Script ift.php completed, returning 0
-- Executing [s@macro-salida-ift:3] Dial("SIP/10001-00000005", "SIP/BLASPavel_TRUNK/5212311143483,,Tor") in new stack
[Jun 5 22:35:40] WARNING[4352][C-00000005]: app_dial.c:2455 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-salida-ift:4] Hangup("SIP/10001-00000005", "") in new stack
== Spawn extension (macro-salida-ift, s, 4) exited non-zero on 'SIP/10001-00000005' in macro 'salida-ift'
== Spawn extension (default, 412311143483, 1) exited non-zero on 'SIP/10001-00000005'
-- Executing [h@default:1] AGI("SIP/10001-00000005", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack


[BLASPavel_TRUNK]
type=friend
host=sipwest.kirasip.com
username=6623849930
secret=riurTtT54eDfHGByj654
port=5060
dtmfmode=rfc2833
qualify=yes
progressinband=never
canreinvite=no
nat=yes
context=from-trunk
disallow=all
allow=alaw
allow=ulaw
allow=G729



The characteristics of my server are the following

8 vcore
8 gb
380

Vicidial version 2.14b0.5

[striker]

make sure your sip trunk registered by running below commands in asterisk cli

sip show peers
sip show registry

also post the Dialplan used to dial out .