Need help

I was running freepbx from svn OK.
Then after checking “Modules Update” I was told that upgrades were
available but that I needed to upgrade to 2.3.0beta which I did.

Now FreePBX is broken.
I’ve double checked permissions. I see nothing wrong.
But when I try to dial internal extensions, I get “unavailable, please leave a message” and outside callers get “the number you have dialed is out of order!!!”

How do I fix this? Can I downgrade?

Here is some debug output:

DIALING <101>

-- Executing Set("SIP/104-081e3c10", "__RINGTIMER=30") in new stack
-- Executing Macro("SIP/104-081e3c10", "exten-vm|101|101") in new stack
-- Executing Macro("SIP/104-081e3c10", "user-callerid") in new stack
-- Executing NoOp("SIP/104-081e3c10", "user-callerid: device 104") in new stack
-- Executing Set("SIP/104-081e3c10", "AMPUSER=104") in new stack
-- Executing GotoIf("SIP/104-081e3c10", "0?report") in new stack
-- Executing GotoIf("SIP/104-081e3c10", "0?start") in new stack
-- Executing Set("SIP/104-081e3c10", "REALCALLERIDNUM=104") in new stack
-- Executing NoOp("SIP/104-081e3c10", "REALCALLERIDNUM is 104") in new stack
-- Executing Set("SIP/104-081e3c10", "AMPUSER=104") in new stack
-- Executing Set("SIP/104-081e3c10", "AMPUSERCIDNAME=USER") in new stack
-- Executing GotoIf("SIP/104-081e3c10", "0?report") in new stack
-- Executing Set("SIP/104-081e3c10", "AMPUSERCID=104") in new stack
-- Executing Set("SIP/104-081e3c10", "CALLERID(all)=USER <104>") in new stack
-- Executing Set("SIP/104-081e3c10", "REALCALLERIDNUM=104") in new stack
-- Executing NoOp("SIP/104-081e3c10", "TTL:  ARG1: 101") in new stack
-- Executing GotoIf("SIP/104-081e3c10", "0?continue") in new stack
-- Executing Set("SIP/104-081e3c10", "__TTL=64") in new stack
-- Executing GotoIf("SIP/104-081e3c10", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing NoOp("SIP/104-081e3c10", "Using CallerID "USER" <104>") in new stack
-- Executing Set("SIP/104-081e3c10", "FROMCONTEXT=exten-vm") in new stack
-- Executing Set("SIP/104-081e3c10", "VMBOX=101") in new stack
-- Executing Set("SIP/104-081e3c10", "EXTTOCALL=101") in new stack
-- Executing Set("SIP/104-081e3c10", "CFUEXT=") in new stack
-- Executing Set("SIP/104-081e3c10", "CFBEXT=") in new stack
-- Executing Set("SIP/104-081e3c10", "RT=30") in new stack
-- Executing Macro("SIP/104-081e3c10", "record-enable|101|IN") in new stack
-- Executing GotoIf("SIP/104-081e3c10", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/104-081e3c10", "recordingcheck|20070709-164931|1184014171.0") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/104-081e3c10", "No recording needed") in new stack
-- Executing Macro("SIP/104-081e3c10", "dial|30|Ttrw|101") in new stack
-- Executing DeadAGI("SIP/104-081e3c10", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
-- AGI Script dialparties.agi completed, returning 0
-- Executing NoOp("SIP/104-081e3c10", "Returned from dialparties with no extensions to call") in new stack
-- Executing NoOp("SIP/104-081e3c10", "DIALSTATUS is ") in new stack
-- Executing Set("SIP/104-081e3c10", "SV_DIALSTATUS=") in new stack
-- Executing GosubIf("SIP/104-081e3c10", "0?docfu|1") in new stack
-- Executing GosubIf("SIP/104-081e3c10", "0?docfb|1") in new stack
-- Executing Set("SIP/104-081e3c10", "DIALSTATUS=") in new stack
-- Executing NoOp("SIP/104-081e3c10", "Voicemail is 101") in new stack
-- Executing GotoIf("SIP/104-081e3c10", "0?s-|1") in new stack
-- Executing NoOp("SIP/104-081e3c10", "Sending to Voicemail box 101") in new stack
-- Executing Macro("SIP/104-081e3c10", "vm|101|") in new stack
-- Executing Macro("SIP/104-081e3c10", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/104-081e3c10", "user-callerid: Jeremy Johnson 104") in new stack
-- Executing Set("SIP/104-081e3c10", "AMPUSER=104") in new stack
-- Executing GotoIf("SIP/104-081e3c10", "0?report") in new stack
-- Executing GotoIf("SIP/104-081e3c10", "1?start") in new stack
-- Goto (macro-user-callerid,s,6)
-- Executing NoOp("SIP/104-081e3c10", "REALCALLERIDNUM is 104") in new stack
-- Executing Set("SIP/104-081e3c10", "AMPUSER=104") in new stack
-- Executing Set("SIP/104-081e3c10", "AMPUSERCIDNAME=USER") in new stack
-- Executing GotoIf("SIP/104-081e3c10", "0?report") in new stack
-- Executing Set("SIP/104-081e3c10", "AMPUSERCID=104") in new stack
-- Executing Set("SIP/104-081e3c10", "CALLERID(all)=USER<104>") in new stack
-- Executing Set("SIP/104-081e3c10", "REALCALLERIDNUM=104") in new stack
-- Executing NoOp("SIP/104-081e3c10", "TTL: 64 ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/104-081e3c10", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing NoOp("SIP/104-081e3c10", "Using CallerID "USER" <104>") in new stack
-- Executing Set("SIP/104-081e3c10", "VMGAIN=") in new stack
-- Executing GotoIf("SIP/104-081e3c10", "1?vmx|1") in new stack
-- Goto (macro-vm,vmx,1)
-- Executing Set("SIP/104-081e3c10", "MODE=unavail") in new stack
-- Executing GotoIf("SIP/104-081e3c10", "1?notdirect") in new stack
-- Goto (macro-vm,vmx,4)
-- Executing NoOp("SIP/104-081e3c10", "Checking if ext 101 is enabled: enabled") in new stack
-- Executing GotoIf("SIP/104-081e3c10", "0?s-|1") in new stack
-- Executing Macro("SIP/104-081e3c10", "get-vmcontext|101") in new stack
-- Executing Set("SIP/104-081e3c10", "VMCONTEXT=default") in new stack
-- Executing GotoIf("SIP/104-081e3c10", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing NoOp("SIP/104-081e3c10", "") in new stack
-- Executing System("SIP/104-081e3c10", "ls /var/spool/asterisk/voicemail/default/101/unavail.wav") in new stack

ls: cannot access /var/spool/asterisk/voicemail/default/101/unavail.wav: No such file or directory
– Executing NoOp(“SIP/104-081e3c10”, “File for mode: unavail does not exist| going to normal voicemail”) in new stack
– Executing Goto(“SIP/104-081e3c10”, “s-|1”) in new stack
– Goto (macro-vm,s-,1)
– Executing Macro(“SIP/104-081e3c10”, “get-vmcontext|101”) in new stack
– Executing Set(“SIP/104-081e3c10”, “VMCONTEXT=default”) in new stack
– Executing GotoIf(“SIP/104-081e3c10”, “0?200:300”) in new stack
– Goto (macro-get-vmcontext,s,300)
– Executing NoOp(“SIP/104-081e3c10”, “”) in new stack
– Executing VoiceMail(“SIP/104-081e3c10”, “[email protected]|u”) in new stack
– Playing ‘/var/spool/asterisk/voicemail/default/101/unavail’ (language ‘en’)
== Spawn extension (macro-vm, s-, 2) exited non-zero on ‘SIP/104-081e3c10’ in macro ‘vm’
== Spawn extension (macro-vm, s-, 2) exited non-zero on ‘SIP/104-081e3c10’ in macro ‘exten-vm’
== Spawn extension (macro-vm, s-, 2) exited non-zero on ‘SIP/104-081e3c10’

DIALING <7777 simulate incoming call>

– Executing Goto(“SIP/104-081e3c10”, “from-pstn|s|1”) in new stack
– Goto (from-pstn,s,1)
– Executing NoOp(“SIP/104-081e3c10”, “No DID or CID Match”) in new stack
– Executing Answer(“SIP/104-081e3c10”, “”) in new stack
– Executing Wait(“SIP/104-081e3c10”, “2”) in new stack
– Executing Playback(“SIP/104-081e3c10”, “ss-noservice”) in new stack :evil:
– Playing ‘ss-noservice’ (language ‘en’)
== Spawn extension (from-pstn, s, 4) exited non-zero on ‘SIP/104-081e3c10’

/var/www/html/admin/modules/core/agi-bin/dialparties.agi

is this the way yours is???

require_once “phpagi.php”;
require_once “phpagi-asmanager.php”;

// Diego Iastrubni [email protected] and the FreePBX community

$config = parse_amportal_conf( “/etc/amportal.conf” );

require_once “phpagi.php”;
require_once “phpagi-asmanager.php”;

$debug = 4;

$ext = array();

#asterisk -vvv
call office using PSTN from outside:

-- Starting simple switch on 'Zap/5-1'
-- Executing NoOp("Zap/5-1", "No DID or CID Match") in new stack
-- Executing Answer("Zap/5-1", "") in new stack
-- Executing Wait("Zap/5-1", "2") in new stack
-- Executing Playback("Zap/5-1", "ss-noservice") in new stack
-- Playing 'ss-noservice' (language 'en')
-- Executing SayAlpha("Zap/5-1", "") in new stack

OK, I read the example for the new Day/Night module.
I setup a default inbound route to point to a Day/Night definition
which in turn points to IVR-Open

I don’t recall whethor I had explicitly setup a default inbound route
before, or whethor the new 2.3beta upgrade deleted the old inbound route setting when creating the new Day/Night module.