My dial by name directory has a problem. When I dial # or 411 the entry greeting doesn’t play. I hear the message no directory entries match your search. I am using asterisk 11.7.0
I have tried a lot of things to try and fix it. could someone show me the files I need to look at to trace the problem in a step by step manner. The app-directory looks right and the pbxdirectory looks right. Where do I find the documentation of what recordings need to be there for this to work.
The quickest way to find out is to watch the asterisk console while you try that call.
Here is the console output when I dial # from a sip phone. 411 does the same. I don’t know what it is telling me. I haven’t modified any agi scripts. I have loaded the Voice Vector prompts. I have also reinstalled the original prompts but get the same result. Voicemail.conf is there but I cannot find a place to tell asterisk to look for the first name or last name.
Connected to Asterisk 11.7.0 currently running on voysbox (pid = 3364)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Extension Changed 5005[ext-local] new state InUse for Notify User 5041
– Executing [#@from-internal:1] Answer(“SIP/5005-000000a2”, “”) in new stack
> 0x2b1d004ebda0 – Probation passed - setting RTP source address to 10.10.204.109:4000
– Executing [#@from-internal:2] Wait(“SIP/5005-000000a2”, “1”) in new stack
> 0x2b1d004ebda0 – Probation passed - setting RTP source address to 10.10.204.109:4000
– Executing [#@from-internal:3] AGI(“SIP/5005-000000a2”, “directory,from-did-direct,bo”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/directory
directory,from-did-direct,bo: Notice: vm-context not specified. Using ‘default’
– Playing ‘dir-nomatch’ (escape_digits=) (sample_offset 0)
– Playing ‘dir-nomatch’ (escape_digits=) (sample_offset 0)
– <SIP/5005-000000a2>AGI Script directory completed, returning 4
== Spawn extension (from-internal, #, 3) exited non-zero on ‘SIP/5005-000000a2’
– Executing [h@from-internal:1] Macro(“SIP/5005-000000a2”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/5005-000000a2”, “1?endmixmoncheck”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] NoOp(“SIP/5005-000000a2”, “End of MIXMON check”) in new stack
– Executing [s@macro-hangupcall:10] GotoIf(“SIP/5005-000000a2”, “1?nomeetmemon”) in new stack
– Goto (macro-hangupcall,s,28)
– Executing [s@macro-hangupcall:28] NoOp(“SIP/5005-000000a2”, “End of MEETME check”) in new stack
– Executing [s@macro-hangupcall:29] GotoIf(“SIP/5005-000000a2”, “1?noautomon”) in new stack
– Goto (macro-hangupcall,s,34)
– Executing [s@macro-hangupcall:34] NoOp(“SIP/5005-000000a2”, “TOUCH_MONITOR_OUTPUT=”) in new stack
– Executing [s@macro-hangupcall:35] GotoIf(“SIP/5005-000000a2”, “1?noautomon2”) in new stack
– Goto (macro-hangupcall,s,41)
– Executing [s@macro-hangupcall:41] NoOp(“SIP/5005-000000a2”, “MONITOR_FILENAME=”) in new stack
– Executing [s@macro-hangupcall:42] GotoIf(“SIP/5005-000000a2”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,45)
– Executing [s@macro-hangupcall:45] GotoIf(“SIP/5005-000000a2”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,48)
– Executing [s@macro-hangupcall:48] GotoIf(“SIP/5005-000000a2”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,50)
– Executing [s@macro-hangupcall:50] AGI(“SIP/5005-000000a2”, “hangup.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
– <SIP/5005-000000a2>AGI Script hangup.agi completed, returning 0
– Executing [s@macro-hangupcall:51] Hangup(“SIP/5005-000000a2”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 51) exited non-zero on ‘SIP/5005-000000a2’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/5005-000000a2’
== Extension Changed 5005[ext-local] new state Idle for Notify User 5041
Generally, when having problems with third party dialplans and AGI scripts you should contact the support channel for the third party, e.g. f for FreePBX use freepbx.org/forums/