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[Solved] IVR sound doesnt play


#1

Hello,

I am running asterisk 1.8 with freepbx 2.10. In FreePBX I have set up some user. Now I have set up a small test IVR, and uploaded a .wav file PCM encoded 8000HZ and 16 bits. But when I call my DID number I get the following error:

    -- Executing [start@test:21]BackGround("SIP/localtelecom-00000001", "test")in new stack

[2013-03-30 15:40:37] WARNING[3272]: file.c:663 ast_openstream_full: File test does not exist in any format
[2013-03-30 15:40:37] WARNING[3272]: file.c:958 ast_streamfile: unable to open test (format 0x4 (ulaw)): No such file or directory
[2013-03-30 15:40:37] WARNING[3272]: pbx.c:10083 pbx_builtin_background: ast_streamfile failed on SIP/localtelecom-00000001 for test
– Executing [start@test:3] WaitExten(“SIP/localtelecom-00000001”, “60”)in new stack

My context in extensions.conf is the following:

[from-localtelecom]
exten => 1000368,1,Goto(test,start,1)

[test]
exten => start,1,Answer()
same => n,Background(test)
same => n,WaitExten(60)
exten => 1,1,Dial(SIP/105,60,tr)
exten => 2,1,Dial(SIP/110,60,tr)

I need help with this error cause I’m stuck right now.

Can anyone help me, thanks in advance.


#2

you can check if the file is compatible by issuing the following command
$file filename.wav
Also check ownership and permissions


#3

Hello thanks for your reply, one question how can i check the ownership and permissions. And where do I need to enter the command $file filename.wav?

Thanks


#4

Sound files for IVR are usually in /var/lib/asterisk/sounds/custom. Go to that directory.
ls -l will list the files in the directory, their ownership and permissions.
Should be owned by asterisk asterisk and 644 for permissions
[root@local custom]# ls -l
total 984
-rw-r–r-- 1 asterisk asterisk 7418 Mar 29 11:53 Directions.wav
[root@local custom]# file Open.wav
Open.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 8000 Hz


#5

Hello I get the following when I do ls -l:

-rw-r–r-- 1 asterisk asterisk 5327406 Mar 29 11:37 test.wav

When I do “file test.wav” it shows me this:

test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 8000 Hz


#6

You need it to be mono. Asterisk won’t play stereo files


#7

Ok, but is there a way to make it mono instead of stereo?


#8

sox test.wav -c 1 tmp.wav;cp -f tmp.wav test.wav


#9

Also, you can’t program IVR’s manually and use FreePBX, at least not the way you did it.

Why not just add the IVR from FreePBX web interface?


#10

Thanks for the reply, I will try this tomorrow, But I have on question with it. Do I need to set up an dialplan to let it work, or do I just need the IVR in the FreePBX interface?


#11

Just the IVR module.


#12

Hello, I tried to call in just with the IVR from the FreePBX webinterface, but know my call is rejected, the error is the following:

[2013-03-31 00:27:52] Notice[3216]: chan_sip.c:23016 handle_request_invite: call from ‘1000368’ (208.64.255.24:5060) to extension ‘1000368’ rejected because extension not found in context ‘from-localtelecom’.

what does this mean?


#13

Hello, I tried this but no luck it stays stereo


#14

Did you just makeup the context from-localtelecom? That’s not a FreePBX context. Extensions should be in from-internal and trunks in from-pstn


#15

Hello, I have fixed the problem. My dialplan looks like this right now:

[from-localtelecom]
exten => 1000368,1,Goto(test,start,1)

[test]
exten => start,1,Answer()
exten => start,2,Set(GLOBAL(sounds_path)=/var/lib/asterisk/sounds/custom)
exten => start,3,Background(${sounds_path}test1)
exten => 1,1,Dial(SIP/105,60,tr)
exten => 2,1,Dial(SIP/110,60,tr)
exten => t,1,Dial(SIP/999,60,tr)

Now it works perfectly, thanks for all the help.


#16

Hi, I am trying out RasPBX (FreePBX on Raspberry Pi)... everything is working except I can't get any system recordings to play. I checked / changed ownership... permissions... directories... confirmed file formats. None of this is working for me.

Files were uploaded through the System Recording's GUI.

Error log looks like this:

<code>

[2013-11-06 20:04:14] VERBOSE[6841][C-00000002] pbx.c: -- Executing [s@ivr-1:9] Set("SIP/SIPAffinity-Proxy1-00000002", "IVR_MSG=custom/2014OGM1") in new stack
[2013-11-06 20:04:14] VERBOSE[6841][C-00000002] pbx.c: -- Executing [s@ivr-1:10] Set("SIP/SIPAffinity-Proxy1-00000002", "TIMEOUT(digit)=3") in new stack
[2013-11-06 20:04:14] VERBOSE[6841][C-00000002] func_timeout.c: -- Digit timeout set to 3.000
[2013-11-06 20:04:14] VERBOSE[6841][C-00000002] pbx.c: -- Executing [s@ivr-1:11] ExecIf("SIP/SIPAffinity-Proxy1-00000002", "1?Background(custom/2014OGM1)") in new stack
[2013-11-06 20:04:14] WARNING[6841][C-00000002] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2013-11-06 20:04:14] WARNING[6841][C-00000002] file.c: Unable to open custom/2014OGM1 (format (g729)): No such file or directory
[2013-11-06 20:04:14] WARNING[6841][C-00000002] pbx.c: ast_streamfile failed on SIP/SIPAffinity-Proxy1-00000002 for custom/2014OGM1
[2013-11-06 20:04:14] VERBOSE[6841][C-00000002] pbx.c: -- Executing [s@ivr-1:12] WaitExten("SIP/SIPAffinity-Proxy1-00000002", "25,") in new stack

</code>

I have use FreePBX previously with a hosted setup (I administered it, but I did not do the installation or the problem solving) so I am relatively experience with the GUI setup (except this interface looks different from what I'm used ).   So, I'm not sure what to do next.

 

For the recod, I setup my Trunks with the "sample code" provided by my trunk provider (copied and pasted) and as I said, Calls in / out and between SIP extensions are working just fine.

 

Thank you for any help.

Jay

 

 

 

 


(system) closed #17