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 [s@from-pstn-toheader: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 [s@from-pstn-toheader: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 [s@from-pstn-toheader: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 [s@from-pstn-toheader: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 [s@from-pstn: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 [s@from-pstn: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 [s@from-pstn: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 [s@from-pstn:4] Wait(“PJSIP/Supreme_Voice-00000004”, “2”) in new stack
37588 [2022-01-03 22:11:37] VERBOSE[23568][C-00000003] pbx.c: Executing [s@from-pstn: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 [h@from-pstn:1] Macro(“PJSIP/Supreme_Voice-00000004”, “hangupcall,”) in new stack
37591 [2022-01-03 22:11:39] VERBOSE[23568][C-00000003] pbx.c: Executing [s@macro-hangupcall: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 [s@macro-hangupcall: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 [s@macro-hangupcall: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 [s@macro-hangupcall: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 [s@macro-hangupcall: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:s@ip;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: 256154392_78623528@ip
CSeq: 793 CANCEL
Content-Length: 0

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

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