Incoming calls sent directly to no service message

I am using incrediblepi, which is IncrediblePBX for the raspberry pi. Asterisk version 13.18.0. Calls between extensions work fine, but incoming and outgoing do not. I am using and have been for 10 years, during which this has mostly worked but I changed both my configuration and my networking setup recently so I probably did something wrong.

My trunk is registered to and the incoming call does get routed to my server; however, the calls always generate a “no service” message and are dropped. I have no idea what is causing this and would be greatful for any assistance. The logs when I make the incoming call:

Connected to Asterisk 13.18.0 currently running on incrediblepi (pid = 1440)
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
       > 0x6e94e868 -- Strict RTP learning after remote address set to:
    -- Executing [4343274640@from-trunk:1] Set("SIP/CroxtonOutTrunk-00000005", "__FROM_DID=4343274640") in new stack
    -- Executing [4343274640@from-trunk:2] NoOp("SIP/CroxtonOutTrunk-00000005", "Received an unknown call with DID set to 4343274640") in new stack
    -- Executing [4343274640@from-trunk:3] Goto("SIP/CroxtonOutTrunk-00000005", "s,a2") in new stack
    -- Goto (from-trunk,s,2)
    -- Executing [s@from-trunk:2] Answer("SIP/CroxtonOutTrunk-00000005", "") in new stack
    -- Executing [s@from-trunk:3] Log("SIP/CroxtonOutTrunk-00000005", "WARNING,Friendly Scanner from") in new stack
[2023-10-21 18:54:04] WARNING[8451][C-00000006]: Ext. s:3 @ from-trunk: Friendly Scanner from
    -- Executing [s@from-trunk:4] Wait("SIP/CroxtonOutTrunk-00000005", "2") in new stack
    -- Executing [s@from-trunk:5] Playback("SIP/CroxtonOutTrunk-00000005", "ss-noservice") in new stack
    -- <SIP/CroxtonOutTrunk-00000005> Playing 'ss-noservice.gsm' (language 'en')
       > 0x6e94e868 -- Strict RTP switching to RTP target address as source
       > 0x6e94e868 -- Strict RTP learning complete - Locking on source address
    -- Executing [h@from-trunk:1] Macro("SIP/CroxtonOutTrunk-00000005", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/CroxtonOutTrunk-00000005", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] Hangup("SIP/CroxtonOutTrunk-00000005", "") in new stack
  == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/CroxtonOutTrunk-00000005' in macro 'hangupcall'
  == Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/CroxtonOutTrunk-00000005'

You should ask IncrediblePBX community. FYI asterisk 13 is EOL since ages, chan_sip is EOL too and removed from latest asterisk versions.

-- Executing [4343274640@from-trunk:3] Goto("SIP/CroxtonOutTrunk-00000005", "s,a2") in new stack
-- Goto (from-trunk,s,2)

Seems strange to me and send incoming calls to s extension priority 2 which does play the noservice

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This looks like FreePBX dialplan. That is not part of Asterisk. You should really get support from the incrediblePBX community, but to the extent that it is based on FreePBX, may be prepared to help.

You need to add a priority 2 match to from-trunk which matches 4343274640, and goes to code that processes the call, or you need to add code ahead of the logged code, which extracts the number you actually want to match from the incoming request. For FreePBX, the former would be done by correctly configuring incoming routes.

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