Dictation *35 not working?

General Help
1

trixbox 2.2.3 / freepbx 2.2.3 / dictation module 1.1.1

Call trace is below:

One thing that I noticed standing out is that /var/lib/asterisk/bin/audio-email.pl doesn’t exist… but it doesn’t even seem to be getting to that part of the context.

[code:1] – Executing NoOp(“SIP/8500-083f1a70”, “Using CallerID “Grandstream” <8500>”) in new stack
– Executing NoOp(“SIP/8500-083f1a70”, “CallerID is 8500”) in new stack
– Executing Set(“SIP/8500-083f1a70”, “DICTENABLED=enabled”) in new stack
– Executing GotoIf(“SIP/8500-083f1a70”, “0?nodict:dictok”) in new stack
– Goto (from-internal,*34,8)
– Executing Dictate(“SIP/8500-083f1a70”, “/var/lib/asterisk/sounds/dictate/8500”) in new stack
– Playing ‘dictate/enter_filename’ (language ‘en’)
– Playing ‘dictate/forhelp’ (language ‘en’)
– Playing ‘dictate/playback_mode’ (language ‘en’)
– Playing ‘dictate/paused’ (language ‘en’)
– Playing ‘dictate/enter_filename’ (language ‘en’)
– Executing Macro(“SIP/8500-083f1a70”, “hangupcall”) in new stack
– Executing ResetCDR(“SIP/8500-083f1a70”, “w”) in new stack
– Executing NoCDR(“SIP/8500-083f1a70”, “”) in new stack
– Executing GotoIf(“SIP/8500-083f1a70”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing GotoIf(“SIP/8500-083f1a70”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing Wait(“SIP/8500-083f1a70”, “5”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/8500-083f1a70’ in macro ‘hangupcall’
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/8500-083f1a70’
– Executing Answer(“SIP/8500-083f1a70”, “”) in new stack
– Executing Macro(“SIP/8500-083f1a70”, “user-callerid|”) in new stack
– Executing NoOp(“SIP/8500-083f1a70”, “user-callerid: device 8500”) in new stack
– Executing GotoIf(“SIP/8500-083f1a70”, “0?report”) in new stack
– Executing GotoIf(“SIP/8500-083f1a70”, “0?start”) in new stack
– Executing Set(“SIP/8500-083f1a70”, “REALCALLERIDNUM=8500”) in new stack
– Executing NoOp(“SIP/8500-083f1a70”, “REALCALLERIDNUM is 8500”) in new stack
– Executing Set(“SIP/8500-083f1a70”, “AMPUSER=8500”) in new stack
– Executing Set(“SIP/8500-083f1a70”, “AMPUSERCIDNAME=Grandstream”) in new stack
– Executing GotoIf(“SIP/8500-083f1a70”, “0?report”) in new stack
– Executing Set(“SIP/8500-083f1a70”, “CALLERID(all)=Grandstream <8500>”) in new stack
– Executing Set(“SIP/8500-083f1a70”, “REALCALLERIDNUM=8500”) in new stack
– Executing NoOp(“SIP/8500-083f1a70”, "TTL: ARG1: ") in new stack
– Executing GotoIf(“SIP/8500-083f1a70”, “0?continue”) in new stack
– Executing Set(“SIP/8500-083f1a70”, “__TTL=64”) in new stack
– Executing GotoIf(“SIP/8500-083f1a70”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,21)
– Executing NoOp(“SIP/8500-083f1a70”, “Using CallerID “Grandstream” <8500>”) in new stack
– Executing NoOp(“SIP/8500-083f1a70”, “CallerID is 8500”) in new stack
– Executing Set(“SIP/8500-083f1a70”, “DICTENABLED=enabled”) in new stack
– Executing GotoIf(“SIP/8500-083f1a70”, “0?nodict:dictok”) in new stack
– Goto (from-internal,*35,8)
– Executing Read(“SIP/8500-083f1a70”, “DICTFILE|enter-filename-short||||”) in new stack
– User disconnected
== Spawn extension (from-internal, *35, 8) exited non-zero on ‘SIP/8500-083f1a70’
– Executing Macro(“SIP/8500-083f1a70”, “hangupcall”) in new stack
– Executing ResetCDR(“SIP/8500-083f1a70”, “w”) in new stack
– Executing NoCDR(“SIP/8500-083f1a70”, “”) in new stack
– Executing GotoIf(“SIP/8500-083f1a70”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing GotoIf(“SIP/8500-083f1a70”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing Wait(“SIP/8500-083f1a70”, “5”) in new stack
– Executing Hangup(“SIP/8500-083f1a70”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on ‘SIP/8500-083f1a70’ in macro ‘hangupcall’
== Spawn extension (macro-hangupcall, s, 10) exited non-zero on ‘SIP/8500-083f1a70’[/code:1]

Any help is much appreciated :slight_smile:

2

It seems that – Executing Read(“SIP/8500-083f1a70”, “DICTFILE|enter-filename-short||||”) in new stack might be an issue…

Is there a version higher than 1.1.1 available? Going through updates in freePBX says that I’m up to date.

3

Anyone with insight today? :slight_smile:

4

the log may tell you more, I’m guessing it will tell you that the file does not exist. This would be consistent with audio-email.pl not being there. All these files (sound files, script file) are auto-linked when you apply configuration settings. If that is not happening, something else is broken in your configuration. It is part of what retrieve_conf does when you press the orange bar.

5

Ah, interestingly enough, the files in /var/www/html/admin/modules/dictate/var/lib/asterisk/sounds/ needed to be copied over to /var/lib/asterisk/sounds

6

there should be no files in /var/www/html/admin/modules/dictate/var/lib/asterisk/sounds. The files should be in /var/www/html/admin/modules/dictate/sounds. From there they will get auto-linked.

How did they get to the place you mentioned? How did you install this? Ah - I just looked down, it’s a trixbox install. I believe at one point trixbox was really screwing with how they install modules. (they seem to have a history of thinking they knew how to do things better than the right way - usually breaking stuff) I don’t think they are doing that any more. However, you probably have other broken things like backup and some others.

You would be best to completely reinstall FreePBX if you had this situation because it clearly indicates that something was very messed up on how your system was installed. Or maybe, not quite as drastically, go through each module and if you see any more with such a path as var/lib/astersk/… under the module, then wipe that out and reinstall that module from the online system. Check backup, callback, followme, ringgroups, pbdirectory and probably some others - can’t recall on 2.2.3 vs. 2.3 what we have going.