Calls to one extension from one extension only go to vmail

Hi, never encountered this before and hoping community can help me with it.

When calling one extension from my own extension (on either desktop or softphone), hear the person is on the phone as my call is sent to voice mail. Meanwhile the person is NOT on the phone. When calling the same extension with another line (on same desktop or softphone) the call is answered normally.

Below is what is seen in CLI when trying to call ext. 730 when calling from my own extension:
Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
– Executing [[email protected]:1] GotoIf(“PJSIP/308-00000030”, “1?ext-local,730,1:followme-check,730,1”) in new stack
– Goto (ext-local,730,1)
– Executing [[email protected]:1] Set(“PJSIP/308-00000030”, “__RINGTIMER=18”) in new stack

[2022-01-30 16:53:51] WARNING[24817][C-0000001a]: func_strings.c:1442 function_eval: EVAL requires an argument: EVAL()

Executing [[email protected]:13] ExecIf(“PJSIP/308-00000030”, “0?Set(DIALSTATUS=CHANUNAVAIL)”) in new stack

Spawn extension (from-internal, 730, 1) exited non-zero on ‘PJSIP/730-00000031’
– PJSIP/730-00000031 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
– Called PJSIP/730/sip:[email protected]:47291;transport=TLS;rinstance=c0505145a5f8bc0a;x-ast-orig-host=192.168.1.29:0
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5

Everyone is busy/congested at this time (1:1/0/0)
Executing [[email protected]:3] VoiceMail(“PJSIP/308-00000030”, “[email protected],sbg(12)”) in new stack
== SRTCP unprotect failed on SSRC 1774651290 because of unable to perform desired validation
– <PJSIP/308-00000030> Playing ‘vm-theperson.ulaw’ (language ‘en’)
== SRTCP unprotect failed on SSRC 1774651290 because of unable to perform desired validation
== SRTCP unprotect failed on SSRC 1774651290 because of unable to perform desired validation
But the softphone being called is NOT in use!

Endpoint: 730/730 Not in use 0 of inf
InAuth: 730-auth/730
Aor: 730 2
Contact: 730/sip:[email protected]:42192;transport 68ac6d3bf0 Avail

Asterisk 18.6.0
Sangoma Linux release 7.8.2003
FreePBX 15.0.17.68

Thanks for the help with this.

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