Inbound route, time condition, extension works, IVR does not

Trying to support an Elastix 2.5 FreePBX 2.11 install. The IVR option hangs up on me. Here’s what the call looks like. I can ring inbound route, time condition, user extension. If I change the time condition to IVR, with any new IVR, any recording I select, IVR automatically hangs up. Seems like IVR is broken.

Connected to Asterisk 11.25.0 currently running on localhost (pid = 2865)
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [[email protected]:1] Set("SIP/Trunk1-0000000e", "__FROM_                                                                             DID=+14057942653") in new stack
    -- Executing [[email protected]:2] Gosub("SIP/Trunk1-0000000e", "app-b                                                                             lacklist-check,s,1()") in new stack
    -- Executing [[email protected]:1] GotoIf("SIP/Trunk1-0000000e", "0?blac                                                                             klisted") in new stack
    -- Executing [[email protected]:2] Set("SIP/Trunk1-0000000e", "CALLED_BL                                                                             ACKLIST=1") in new stack
    -- Executing [[email protected]:3] Return("SIP/Trunk1-0000000e", "") in                                                                              new stack
    -- Executing [[email protected]:3] Set("SIP/Trunk1-0000000e", "CDR(did                                                                             )=+14057942653") in new stack
    -- Executing [[email protected]:4] ExecIf("SIP/Trunk1-0000000e", "0 ?S                                                                             et(CALLERID(name)=+14054927711)") in new stack
    -- Executing [[email protected]:5] Set("SIP/Trunk1-0000000e", "CHANNEL                                                                             (musicclass)=default") in new stack
    -- Executing [[email protected]:6] Set("SIP/Trunk1-0000000e", "__MOHCL                                                                             ASS=default") in new stack
    -- Executing [[email protected]:7] Set("SIP/Trunk1-0000000e", "__CALLI                                                                             NGPRES_SV=allowed_not_screened") in new stack
    -- Executing [[email protected]:8] Set("SIP/Trunk1-0000000e", "CALLERP                                                                             RES()=allowed_not_screened") in new stack
    -- Executing [[email protected]:9] Goto("SIP/Trunk1-0000000e", "timeco                                                                             nditions,1,1") in new stack
    -- Goto (timeconditions,1,1)
    -- Executing [[email protected]:1] GotoIfTime("SIP/Trunk1-0000000e", "09:00-1                                                                             7:00,mon-fri,1-31,jan-dec?truestate") in new stack
    -- Goto (timeconditions,1,10)
    -- Executing [[email protected]:10] GotoIf("SIP/Trunk1-0000000e", "0?falsegot                                                                             o") in new stack
    -- Executing [[email protected]:11] ExecIf("SIP/Trunk1-0000000e", "0?Set(DB(T                                                                             C/1)=)") in new stack
    -- Executing [[email protected]:12] Set("SIP/Trunk1-0000000e", "DEVICE_STATE(                                                                             Custom:TC1)=NOT_INUSE") in new stack
    -- Executing [[email protected]:13] ExecIf("SIP/Trunk1-0000000e", "0?Set(DEVI                                                                             CE_STATE(Custom:TCSTICKY)=INUSE)") in new stack
    -- Executing [[email protected]:14] GotoIf("SIP/Trunk1-0000000e", "1?ivr-1,s,                                                                             1") in new stack
    -- Goto (ivr-1,s,1)
    -- Executing [[email protected]:1] Set("SIP/Trunk1-0000000e", "TIMEOUT_LOOPCOUNT=0") i                                                                             n new stack
    -- Executing [[email protected]:2] Set("SIP/Trunk1-0000000e", "INVALID_LOOPCOUNT=0") i                                                                             n new stack
    -- Executing [[email protected]:3] Set("SIP/Trunk1-0000000e", "_IVR_CONTEXT_ivr-1=") i                                                                             n new stack
    -- Executing [[email protected]:4] Set("SIP/Trunk1-0000000e", "_IVR_CONTEXT=ivr-1") in                                                                              new stack
    -- Executing [[email protected]:5] Set("SIP/Trunk1-0000000e", "__IVR_RETVM=") in new s                                                                             tack
    -- Executing [[email protected]:6] GotoIf("SIP/Trunk1-0000000e", "0?skip") in new stac                                                                             k
    -- Executing [[email protected]:7] Answer("SIP/Trunk1-0000000e", "") in new stack
    -- Executing [[email protected]:8] Wait("SIP/Trunk1-0000000e", "1") in new stack
    -- Executing [[email protected]:9] Set("SIP/Trunk1-0000000e", "IVR_MSG=") in new stack
    -- Executing [[email protected]:10] Set("SIP/Trunk1-0000000e", "TIMEOUT(digit)=3") in                                                                              new stack
    -- Digit timeout set to 3.000
    -- Executing [[email protected]:11] ExecIf("SIP/Trunk1-0000000e", "0?Background()") in                                                                              new stack
    -- Executing [[email protected]:12] WaitExten("SIP/Trunk1-0000000e", "20,") in new sta                                                                             ck
[2017-07-17 12:20:35] WARNING[3032]: chan_sip.c:4038 retrans_pkt: Retransmission                                                                              timeout reached on transmission [email protected] for seqno 91291                                                                             7 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retr                                                                             ansmissions
Packet timed out after 6399ms with no response
[2017-07-17 12:20:35] WARNING[3032]: chan_sip.c:4067 retrans_pkt: Hanging up cal                                                                             l [email protected] - no reply to our critical packet (see https:/                                                                             /wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Spawn extension (ivr-1, s, 12) exited non-zero on 'SIP/Trunk1-0000000e'
    -- Executing [[email protected]:1] Hangup("SIP/Trunk1-0000000e", "") in new stack
  == Spawn extension (ivr-1, h, 1) exited non-zero on 'SIP/Trunk1-0000000e'
    -- Registered SIP '3012' at 10.17.1.79:5060
       > Saved useragent "PolycomSoundPointIP-SPIP_550-UA/4.0.4.2906" for peer 3                                                                             012
[2017-07-17 12:20:46] NOTICE[3032]: chan_sip.c:23894 handle_response_peerpoke: P                                                                             eer '3012' is now Reachable. (15ms / 2000ms)
localhost*CLI> exit

If I was troubleshooting, I’d probably start there rather than assume it’s something that’s working for thousands of other people…