Asterisk 16 w/ PJSIP - "Everyone is busy/congested" When Forwarding Inbound Call

Question

I am running Asterisk 16 on CentOS 7 and PJSIP. I am attempting to forward all inbound calls to a phone number (represented by 18001112222) to my cellphone (represented by 12224446666). Registration to the provider and the inbound call getting to the PBX both work without issue, but creating the outbound, forwarded call to my cellphone results in these log lines:

  == Setting global variable 'SIPDOMAIN' to '167.71.106.189'
    -- Executing [18001112222@from-external:1] Verbose("PJSIP/mytrunk-00000000", "1, "New caller, 2224446666 dialed into the IVR."") in new stack
  "New caller, 2224446666 dialed into the IVR."
    -- Executing [18001112222@from-external:2] Answer("PJSIP/mytrunk-00000000", "") in new stack
    -- Executing [18001112222@from-external:3] Dial("PJSIP/mytrunk-00000000", "PJSIP/mytrunk/sip:14045964207") in new stack
    -- Called PJSIP/mytrunk/sip:14045964207
    -- PJSIP/mytrunk-00000000 requested media update control 26, passing it to PJSIP/mytrunk-00000001
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [18001112222@from-external:4] Hangup("PJSIP/mytrunk-00000000", "") in new stack
  == Spawn extension (from-external, 18001112222, 4) exited non-zero on 'PJSIP/mytrunk-00000000'

It would appear that Everyone is busy/congested at this time is the error message causing this setup to fail. I have tried changing the authentication scheme for mytrunk using numerous online forum posts, but the error message continues to show up on the outbound leg of the forward. My configuration files are below:

pjsip.conf:

;==============TRANSPORTS

[simpletrans]
type=transport
protocol=udp
bind=0.0.0.0

;===============TRUNK

[mytrunk]
type=registration
outbound_auth=mytrunk
server_uri=sip:<SIP provider gateway>
client_uri=sip:<username>@<SIP provider gateway>
retry_interval=60

[mytrunk]
type=auth
auth_type=userpass
password=<password>
username=<username>

[mytrunk]
type=aor
contact=sip:<SIP provider gateway>:5060

[mytrunk]
type=endpoint
context=from-external
disallow=all
allow=ulaw
outbound_auth=mytrunk
aors=mytrunk

[mytrunk]
type=identify
endpoint=mytrunk
match=<SIP provider gateway>

extensions.conf:

[globals]
; General internal dialing options used in context Dial-Users.
; Only the timeout is defined here. See the Dial app documentation for
; additional options.
INTERNAL_DIAL_OPT=,30

[from-external]
exten = _18001112222,1,Verbose(1, "New caller, ${CALLERID(num)} dialed in.")
 same = n,Answer()
 same = n,Dial(PJSIP/mytrunk/sip:12224446666@mytrunk)
 same = n,Hangup()

I believe the issue is with the pjsip.conf configuration for outbound calls, because the inbound call reaches the PBX without issue and I am able to do something like Verbose() and see the output in the Asterisk CLI. I am not sure what I need to do in order to create the outbound leg while an inbound call from the ITSP succeeds.

Thank you.

Answer 1

Instead of

same = n,Dial(PJSIP/mytrunk/sip:12224446666@mytrunk)

Try

same = n,Dial(PJSIP/12224446666@mytrunk)

Because all the configuration is in the trunk declaration, you don't need to send the uri to the trunk.