Asterisk outgoing SIP call test

console-output.txt

[Feb 11 13:01:30] WARNING[17454]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/hello-world.call: Operation not permitted
    -- Attempting call on SIP/localhost/sip:17771234567@callcentric.com for application Playback(hello-world) (Retry 1)
  == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
    -- Executing [sip:17771234567@callcentric.com@public:1] Dial("SIP/127.0.0.1-0000000c", "SIP/17771234567@callcentric.com") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/17771234567@callcentric.com
[Feb 11 13:01:30] NOTICE[17430][C-00000013]: chan_sip.c:22978 handle_response_invite: Failed to authenticate on INVITE to '"asterisk" <sip:asterisk@192.168.1.136>;tag=as16ae76cd'
    -- SIP/callcentric.com-0000000d is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Auto fallthrough, channel 'SIP/127.0.0.1-0000000c' status is 'CONGESTION'
    -- Got SIP response 503 "Service Unavailable" back from 127.0.0.1:5060
       > Channel SIP/localhost-0000000b was never answered.
[Feb 11 13:01:30] NOTICE[17498]: pbx_spool.c:389 attempt_thread: Call failed to go through, reason (8) Congestion (circuits busy)
[Feb 11 13:01:30] NOTICE[17498]: pbx_spool.c:392 attempt_thread: Queued call to SIP/localhost/sip:17771234567@callcentric.com expired without completion after 0 attempts

extensions.conf

[public]

include => demo
exten => _[0-9a-zA-Z].,1,Dial(SIP/17771234567@callcentric.com)

hello-world.call

Channel: SIP/localhost/sip:17771234567@callcentric.com
Application: Playback
Data: hello-world

move-hello-world.call

sudo mv hello-world.call /var/spool/asterisk/outgoing/