Internal calls not working

Hi,
I’ve got a few pjsip extensions registered using freepbx.
Here’s the (partial) output from pjsip show endpoints:

 Endpoint:  100/100                                              Not in use    0 of inf
     InAuth:  100-auth/100
        Aor:  100                                                1
      Contact:  100/sip:[email protected]:12410;transport=T 127383ff5c Avail        17.425

 Endpoint:  101/101                                              Not in use    0 of inf
     InAuth:  101-auth/101
        Aor:  101                                                1
      Contact:  101/sip:[email protected]:37249;transport=T c01a62a4d0 Avail         3.575

For some reason, internal calls do not work.

I have so far:

  • checked that both extens use the context from-internal
  • changed my outbound route dialplan from X. to XXXX., because previously these internal calls would be routed out via a trunk
  • tried calling both 101 and (with prefix)*101

I can see in asterisk that the following dialplan function is triggered:

-- Executing [[email protected]:5] Playback("PJSIP/101-0000000f", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack

I’d appreciate any ideas how to fix this.

Can you provide the full call trace of a failed call via pastebin?
https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs-PartII

Here you go:

[2020-06-05 16:41:31] VERBOSE[25494] pbx_variables.c: Setting global variable 'SIPDOMAIN' to 'asterisk.xxx.org'
[2020-06-05 16:41:31] VERBOSE[25494] netsock2.c: Using SIP RTP Audio TOS bits 184
[2020-06-05 16:41:31] VERBOSE[25494] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
[2020-06-05 16:41:31] VERBOSE[25494] netsock2.c: Using SIP RTP Audio CoS mark 5
[2020-06-05 16:41:31] VERBOSE[25732][C-00000002] pbx.c: Executing [[email protected]:1] ResetCDR("PJSIP/100-00000001", "") in new stack
[2020-06-05 16:41:31] VERBOSE[25732][C-00000002] pbx.c: Executing [[email protected]:2] NoCDR("PJSIP/100-00000001", "") in new stack
[2020-06-05 16:41:31] VERBOSE[25732][C-00000002] pbx.c: Executing [[email protected]:3] Progress("PJSIP/100-00000001", "") in new stack
[2020-06-05 16:41:31] VERBOSE[25732][C-00000002] pbx.c: Executing [[email protected]:4] Wait("PJSIP/100-00000001", "1") in new stack
[2020-06-05 16:41:32] VERBOSE[25732][C-00000002] pbx.c: Executing [[email protected]:5] Playback("PJSIP/100-00000001", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[2020-06-05 16:41:32] VERBOSE[25732][C-00000002] file.c: <PJSIP/100-00000001> Playing 'silence/1.slin' (language 'de_DE')
[2020-06-05 16:41:33] VERBOSE[25732][C-00000002] file.c: <PJSIP/100-00000001> Playing 'cannot-complete-as-dialed.slin' (language 'de_DE')
[2020-06-05 16:41:35] VERBOSE[25732][C-00000002] file.c: <PJSIP/100-00000001> Playing 'check-number-dial-again.slin' (language 'de_DE')
[2020-06-05 16:41:37] VERBOSE[25732][C-00000002] pbx.c: Executing [[email protected]:6] Wait("PJSIP/100-00000001", "1") in new stack
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Executing [[email protected]:7] Congestion("PJSIP/100-00000001", "20") in new stack
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Spawn extension (from-internal, 101, 7) exited non-zero on 'PJSIP/100-00000001'
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Executing [[email protected]:1] Macro("PJSIP/100-00000001", "hangupcall") in new stack
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Executing [[email protected]:1] GotoIf("PJSIP/100-00000001", "1?theend") in new stack
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Executing [[email protected]:3] ExecIf("PJSIP/100-00000001", "0?Set(CDR(recordingfile)=)") in new stack
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Executing [[email protected]:4] NoOp("PJSIP/100-00000001", " montior file= ") in new stack
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Executing [[email protected]:5] GotoIf("PJSIP/100-00000001", "1?skipagi") in new stack
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Executing [[email protected]:7] Hangup("PJSIP/100-00000001", "") in new stack
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on 'PJSIP/100-00000001' in macro 'hangupcall'
[2020-06-05 16:41:38] VERBOSE[25732][C-00000002] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/100-00000001'

How odd, there is no associated dialplan for your local extensions. How were they created? If you edit an extension, submit the page with no changes and apply config, does anything change?

From my research, only the users extensions are added to the dialplan (specifically, the contextext-local) when using freepbx in users/devices mode. The devices are not added to the dialplan.

Therefore it’s necessary to add them to extensions_custom.conf manually, for example like this:

[from-internal-custom]
exten => 100,1,Dial(PJSIP/100)
exten => 101,1,Dial(PJSIP/101)

This could possibly be automated through a module.

Does such a module exist (or perhaps, even a setting somewhere in freepbx)?

I’m sure you know, but U&D is unsupported. Every single request for help needs to start out with this fact.

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