From-pstn-toheader not passing the DID from the header

When using the from-trunk-toheader context it still doesn’t pass the did info

the provider uses the header to pass the DID

anyone get Ringlogix to work with pjsip?

‘s’ was rejected because it was not found in from-pstn-toheader context

Logs ?

37574 [2022-01-03 22:11:35] VERBOSE[23568][C-00000003] pbx.c: Executing [[email protected]:1] NoOp(“PJSIP/Supreme_Voice-00000004”, “Attempting to extract DID from SIP To header”) in new stack
37575 [2022-01-03 22:11:35] VERBOSE[23568][C-00000003] pbx.c: Executing [[email protected]:2] GotoIf(“PJSIP/Supreme_Voice-00000004”, “0?SIP”) in new stack
37576 [2022-01-03 22:11:35] VERBOSE[23568][C-00000003] pbx.c: Executing [[email protected]:3] GotoIf(“PJSIP/Supreme_Voice-00000004”, “1?PJSIP”) in new stack
37577 [2022-01-03 22:11:35] VERBOSE[23568][C-00000003] pbx_builtins.c: Goto (from-pstn-toheader,s,7)
37578 [2022-01-03 22:11:35] ERROR[23568][C-00000003] func_cut.c: Syntax: CUT(,,) - missing argument!
37579 [2022-01-03 22:11:35] VERBOSE[23568][C-00000003] pbx.c: Executing [[email protected]:7] Goto(“PJSIP/Supreme_Voice-00000004”, “from-pstn,1”) in new stack
37580 [2022-01-03 22:11:35] VERBOSE[23568][C-00000003] pbx_builtins.c: Goto (from-pstn,s,1)
37581 [2022-01-03 22:11:35] VERBOSE[23568][C-00000003] pbx.c: Executing [[email protected]:1] NoOp(“PJSIP/Supreme_Voice-00000004”, “No DID or CID Match”) in new stack
37582 [2022-01-03 22:11:35] VERBOSE[23568][C-00000003] pbx.c: Executing [[email protected]:2] Answer(“PJSIP/Supreme_Voice-00000004”, “”) in new stack
37583 [2022-01-03 22:11:35] ERROR[23568][C-00000003] pbx_functions.c: Function SIP_HEADER not registered
37584 [2022-01-03 22:11:35] VERBOSE[23568][C-00000003] pbx.c: Executing [[email protected]:3] Log(“PJSIP/Supreme_Voice-00000004”, "WARNING,Friendly Scanner from ") in new stack
37585 [2022-01-03 22:11:35] WARNING[23568][C-00000003] Ext. s: Friendly Scanner from
37586 [2022-01-03 22:11:35] VERBOSE[23568][C-00000003] pbx.c: Executing [[email protected]:4] Wait(“PJSIP/Supreme_Voice-00000004”, “2”) in new stack
37588 [2022-01-03 22:11:37] VERBOSE[23568][C-00000003] pbx.c: Executing [[email protected]:5] Playback(“PJSIP/Supreme_Voice-00000004”, “ss-noservice”) in new stack
37589 [2022-01-03 22:11:37] VERBOSE[23568][C-00000003] file.c: <PJSIP/Supreme_Voice-00000004> Playing ‘ss-noservice.ulaw’ (language ‘en’)
37590 [2022-01-03 22:11:39] VERBOSE[23568][C-00000003] pbx.c: Executing [[email protected]:1] Macro(“PJSIP/Supreme_Voice-00000004”, “hangupcall,”) in new stack
37591 [2022-01-03 22:11:39] VERBOSE[23568][C-00000003] pbx.c: Executing [[email protected]:1] GotoIf(“PJSIP/Supreme_Voice-00000004”, “1?theend”) in new stack
37592 [2022-01-03 22:11:39] VERBOSE[23568][C-00000003] pbx_builtins.c: Goto (macro-hangupcall,s,3)
37593 [2022-01-03 22:11:39] VERBOSE[23568][C-00000003] pbx.c: Executing [[email protected]:3] ExecIf(“PJSIP/Supreme_Voice-00000004”, “0?Set(CDR(recordingfile)=)”) in new stack
37594 [2022-01-03 22:11:39] VERBOSE[23568][C-00000003] pbx.c: Executing [[email protected]:4] NoOp(“PJSIP/Supreme_Voice-00000004”, " montior file= ") in new stack
37595 [2022-01-03 22:11:39] VERBOSE[23568][C-00000003] pbx.c: Executing [[email protected]:5] GotoIf(“PJSIP/Supreme_Voice-00000004”, “1?skipagi”) in new stack
37596 [2022-01-03 22:11:39] VERBOSE[23568][C-00000003] pbx_builtins.c: Goto (macro-hangupcall,s,7)
37597 [2022-01-03 22:11:39] VERBOSE[23568][C-00000003] pbx.c: Executing [[email protected]:7] Hangup(“PJSIP/Supreme_Voice-00000004”, “”) in new stack
37598 [2022-01-03 22:11:39] VERBOSE[23568][C-00000003] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/Supreme_Voice-00000004’ in macro ‘hangupcall’
37599 [2022-01-03 22:11:39] VERBOSE[23568][C-00000003] pbx.c: Spawn extension (from-pstn, h, 1) exited non-zero on ‘PJSIP/Supreme_Voice-00000004’

OK, so what does the To header actually contain?

How do I check that?

“pjsip set logger on” as a CLI command, then look at the log.

<— Received SIP request (389 bytes) from UDP:192.92.8.30:5060 —>
CANCEL sip:[email protected];transport=UDP;line=cnathni SIP/2.0
Via: SIP/2.0/UDP 192.92.8.30:5060;branch=z9hG4bK-524287-1-Y2FhZGVmNmNmZmUyNThlODA1YzI3NGYzOGE0ZDUxOWM.–567ef0290c972559;rport
Max-Forwards: 70
To: sip:[email protected]
From: sip:[email protected];tag=sbieuz4ervc7az7w.o
Call-ID: [email protected]
CSeq: 793 CANCEL
Content-Length: 0

I ran fwconsole refreshsignatures redownload all the config files and it started working.