Hi,
I’m having an issue with an IVR on my FreePBX (5.211.65-13) install. Until this morning everything’s been working just fine, but since I had to reboot a few hours ago we have no audio on our IVR. I’m not sure if it’s a NAT issue, DIDs that dial extensions directly still work. If I set the destination for the main DID to a ring group or specific extension, it works just fine. If it’s set to the IVR, then there is no audio either way. I re-checked firewall rules (even though no IPs or configurations changed, everything’s static) and checked the fail2ban lists on the PBX just in case.
This is the log output when someone calls in to the IVR currently:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [***********@from-trunk:1] Set("SIP/Flowroute-LA-0000001a", "__FROM_DID=***********") in new stack
-- Executing [***********@from-trunk:2] Set("SIP/Flowroute-LA-0000001a", "CDR(did)=***********") in new stack
-- Executing [***********@from-trunk:3] ExecIf("SIP/Flowroute-LA-0000001a", "1 ?Set(CALLERID(name)=+***********)") in new stack
-- Executing [***********@from-trunk:4] Set("SIP/Flowroute-LA-0000001a", "CHANNEL(musicclass)=default") in new stack
-- Executing [***********@from-trunk:5] Set("SIP/Flowroute-LA-0000001a", "__MOHCLASS=default") in new stack
-- Executing [***********@from-trunk:6] Set("SIP/Flowroute-LA-0000001a", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [***********@from-trunk:7] Set("SIP/Flowroute-LA-0000001a", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing [***********@from-trunk:8] Set("SIP/Flowroute-LA-0000001a", "__ALERT_INFO=MAIN") in new stack
-- Executing [***********@from-trunk:9] Goto("SIP/Flowroute-LA-0000001a", "ivr-2,s,1") in new stack
-- Goto (ivr-2,s,1)
-- Executing [s@ivr-2:1] Set("SIP/Flowroute-LA-0000001a", "INVALID_LOOPCOUNT=0") in new stack
-- Executing [s@ivr-2:2] Set("SIP/Flowroute-LA-0000001a", "_IVR_CONTEXT_ivr-2=") in new stack
-- Executing [s@ivr-2:3] Set("SIP/Flowroute-LA-0000001a", "_IVR_CONTEXT=ivr-2") in new stack
-- Executing [s@ivr-2:4] Set("SIP/Flowroute-LA-0000001a", "__IVR_RETVM=") in new stack
-- Executing [s@ivr-2:5] GotoIf("SIP/Flowroute-LA-0000001a", "0?skip") in new stack
-- Executing [s@ivr-2:6] Answer("SIP/Flowroute-LA-0000001a", "") in new stack
-- Executing [s@ivr-2:7] Wait("SIP/Flowroute-LA-0000001a", "1") in new stack
-- Executing [s@ivr-2:8] Set("SIP/Flowroute-LA-0000001a", "IVR_MSG=custom/NewIntro") in new stack
-- Executing [s@ivr-2:9] Set("SIP/Flowroute-LA-0000001a", "TIMEOUT(digit)=3") in new stack
-- Digit timeout set to 3.000
-- Executing [s@ivr-2:10] ExecIf("SIP/Flowroute-LA-0000001a", "1?Background(custom/NewIntro)") in new stack
-- <SIP/Flowroute-LA-0000001a> Playing 'custom/NewIntro.slin' (language 'en')
== Spawn extension (ivr-2, s, 10) exited non-zero on 'SIP/Flowroute-LA-0000001a'
It’s just odd to me that a reboot would cause this with no configuration changes since the last reboot. Am I completely missing something here? It’s like RTP is forwarding properly to the phones but not the PBX itself. Thanks in advance for any help!