Issues with installing on FreePBX - Could not find asterisk.h
milutin-b opened this issue · 6 comments
Hi there.
I have a FPBX server installed from ISO image downloaded from official site working just fine. But have issues following steps outlined in Readme though with ./configure command spitting out Could not find asterisk.h, make sure Asterisk development package is installed error.
I've installed dev package with yum install asterisk18-devel and now can find asterisk.h in /usr/include directory but still get the same error.
My guess is I'm using ./configure command with wrong parameters but I'm unsure what those should be.
I'm confused with provided example - --with-asterisk=<path_to_asterisk_source> --prefix=<path_to_install>. What should those represent? Found few pages like this one but it's still unclear what the actual parametrs should be. I tried few directories as parameters, like: /usr/src/freepbx, /var/lib/asterisk, etc... but now it's just pushing buttons randomly.
Anyone has any idea what am I doing wrong?
./configure --with-asterisk=/usr --prefix=/usr
should work for you
Hi Nickolay thanks for answering.
That worked thank you so much.
I have a different problem now. After running the ./configure one of the three modules isn't working.
ERROR[24515] loader.c: Error loading module 'res_speech_vosk.so': /usr/lib64/asterisk/modules/res_speech_vosk.so: cannot open shared object file: No such file or directory
When running asterisk -rx "module show like res_speech.so" and asterisk -rx "module show like res_http_websocket.so" both of them seem operational.
I am installing Vosk on the same machine FreePBX is running on.
Should I open another issue and close this one?
You need to open file manager or just a terminal and find that so file. It is probably installed in some other location, not /usr/lib64/asterisk/modules, most likely /usr/lib without 64
It really was in /usr/lib/asterisk/modules/ copied it, reloaded and it works.
Thank you.
I have one more question if you would.
I can't seem to pick up conversation using config code from Readme file.
I edited extension_custom.conf to be:
[from-internal-custom]
exten => 100,1,Answer
same => n,Wait(1)
same => n,Dial(PJSIP/${EXTEN})
same => n,Noop(Call is answered)
same => n,SpeechCreate
same => n,SpeechBackground(splat)
same => n,Verbose(0,Result was ${SPEECH_TEXT(0)})
and can only see the transcription from sound file I provided in SpeechBackground as an argument.
If I comment out the Dial line in the conf file the transcription works unidirectiona - that is I can, for a short while, read what I as a caller am saying. But then the line just breaks.
Is there some other function I should be calling or some other way to capture the Vosk output via script etc?
Thanks for any help.
SpeechBackground is only useful for IVR. If you want to monitor the call you have to record with MixMonitor and then transcribe with a separate python script.
Thank you Nickolay