Hello I’m having issues with outbound calls.
I get a message stating “All circuits are busy”
this is my first FreePBX install or ANY Phone server for that matter.
Any help would be greatly appreciated.
thank you in advance
CLI and SIP settings pasted below
type=peer
host=sip.digiumcloud.net
defaultuser=username
fromuser=username
secret=password
insecure=invite
trustrpid=yes
sendrpid=pai
context=from-digium-siptrunk
directmedia=no
disallow=all
allow=g722&ulaw&g729
Connected to Asterisk 16.6.2 currently running on freepbx (pid = 40539)
[Dec 31 11:14:27] == Using SIP RTP TOS bits 184
[Dec 31 11:14:27] == Using SIP RTP CoS mark 5
[Dec 31 11:14:27] > 0x7f5eec030320 – Strict RTP learning after remote address set to: 192.168.4.89:63912
[Dec 31 11:14:27] – Executing [916199547723@from-internal:1] ResetCDR("SIP/112-00000041", "") in new stack
[Dec 31 11:14:27] – Executing [916199547723@from-internal:2] NoCDR("SIP/112-00000041", "") in new stack
[Dec 31 11:14:27] – Executing [916199547723@from-internal:3] Progress("SIP/112-00000041", "") in new stack
[Dec 31 11:14:27] – Executing [916199547723@from-internal:4] Wait("SIP/112-00000041", "1") in new stack
[Dec 31 11:14:27] > 0x7f5eec030320 – Strict RTP switching to RTP target address 192.168.4.89:63912 as source
[Dec 31 11:14:28] – Executing [916199547723@from-internal:5] Playback("SIP/112-00000041", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[Dec 31 11:14:28] – <SIP/112-00000041> Playing ‘silence/1.ulaw’ (language ‘en’)
[Dec 31 11:14:29] – <SIP/112-00000041> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
[Dec 31 11:14:31] – <SIP/112-00000041> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)
[Dec 31 11:14:32] > 0x7f5eec030320 – Strict RTP learning complete - Locking on source address 192.168.4.89:63912
[Dec 31 11:14:34] – Executing [916199547723@from-internal:6] Wait("SIP/112-00000041", "1") in new stack
[Dec 31 11:14:35] – Executing [916199547723@from-internal:7] Congestion("SIP/112-00000041", "20") in new stack
[2019-12-31 11:14:35] WARNING[56219][C-00000029]: channel.c:4864 ast_prod: Prodding channel ‘SIP/112-00000041’ failed
[Dec 31 11:14:35] == Spawn extension (from-internal, 916199547723, 7) exited non-zero on ‘SIP/112-00000041’
[Dec 31 11:14:35] – Executing [h@from-internal:1] Macro("SIP/112-00000041", "hangupcall") in new stack
[Dec 31 11:14:35] – Executing [s@macro-hangupcall:1] GotoIf("SIP/112-00000041", "1?theend") in new stack
[Dec 31 11:14:35] – Goto (macro-hangupcall,s,3)
[Dec 31 11:14:35] – Executing [s@macro-hangupcall:3] ExecIf("SIP/112-00000041", "0?Set(CDR(recordingfile)=)") in new stack
[Dec 31 11:14:35] – Executing [s@macro-hangupcall:4] NoOp("SIP/112-00000041", " montior file= ") in new stack
[Dec 31 11:14:35] – Executing [s@macro-hangupcall:5] GotoIf("SIP/112-00000041", "1?skipagi") in new stack
[Dec 31 11:14:35] – Goto (macro-hangupcall,s,7)
[Dec 31 11:14:35] – Executing [s@macro-hangupcall:7] Hangup("SIP/112-00000041", "") in new stack
[Dec 31 11:14:35] == Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘SIP/112-00000041’ in macro ‘hangupcall’
[Dec 31 11:14:35] == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/112-00000041’
freepbx*CLI>