Custom audio files don't work anymore

mannana93 · May 23, 2015, 8:49am

Hi,
after some time my IVR doesn’t work anymore.
When I call I don’t listen anything, it’s mute. Checking the log I found this:

[2015-05-22 19:02:47] WARNING[6161] file.c: File custom/ivr_rec_ok does not exist in any format
[2015-05-22 19:02:47] WARNING[6161] file.c: Unable to open custom/ivr_rec_ok (format 0x8 (alaw)): No such file or directory
[2015-05-22 19:02:47] WARNING[6161] pbx.c: ast_streamfile failed on SIP/SIP-0xxxxxxx-00000005 for custom/ivr_rec_ok
[2015-05-22 19:02:52] WARNING[6161] file.c: File no-valid-responce-pls-try-again does not exist in any format
[2015-05-22 19:02:52] WARNING[6161] file.c: Unable to open no-valid-responce-pls-try-again (format 0x8 (alaw)): No such file or directory
[2015-05-22 19:02:52] WARNING[6161] pbx.c: ast_streamfile failed on SIP/SIP-0xxxxxx-00000005 for no-valid-responce-pls-try-again

Overkill · May 23, 2015, 9:36am

Either asterisk doesn’t have permissions (amportal chown) or the files don’t exist in a codec that you have enabled on your PBX.

mannana93 · May 23, 2015, 9:48am

The files are in their respective locations, so how I can set correctly permissions?

mannana93 · May 23, 2015, 10:06am

I executed “sudo amportal chown” and restarted asterisk but the problem is still here (the music on hold works correctly).

[2015-05-23 12:02:25] VERBOSE[9102] netsock2.c: == Using SIP RTP TOS bits 184
[2015-05-23 12:02:25] VERBOSE[9102] netsock2.c: == Using SIP RTP CoS mark 5
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [0xxxxxx38@from-trunk-sip-SIP-0xxxxxx38:1] Set("SIP/SIP-0xxxxxx38-00000001", "GROUP()=OUT_2") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [0xxxxxx38@from-trunk-sip-SIP-0xxxxxx38:2] Goto("SIP/SIP-0xxxxxx38-00000001", "from-trunk,0xxxxxx38,1") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Goto (from-trunk,0xxxxxx38,1)
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [0xxxxxx38@from-trunk:1] Set("SIP/SIP-0xxxxxx38-00000001", "__FROM_DID=0xxxxxx38") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [0xxxxxx38@from-trunk:2] Gosub("SIP/SIP-0xxxxxx38-00000001", "app-blacklist-check,s,1()") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [s@app-blacklist-check:1] GotoIf("SIP/SIP-0xxxxxx38-00000001", "0?blacklisted") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [s@app-blacklist-check:2] Set("SIP/SIP-0xxxxxx38-00000001", "CALLED_BLACKLIST=1") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [s@app-blacklist-check:3] Return("SIP/SIP-0xxxxxx38-00000001", "") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [0xxxxxx38@from-trunk:3] Set("SIP/SIP-0xxxxxx38-00000001", "CDR(did)=0xxxxxx38") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [0xxxxxx38@from-trunk:4] ExecIf("SIP/SIP-0xxxxxx38-00000001", "1 ?Set(CALLERID(name)=3xxxxxxxx9)") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [0xxxxxx38@from-trunk:5] Set("SIP/SIP-0xxxxxx38-00000001", "CHANNEL(musicclass)=default") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [0xxxxxx38@from-trunk:6] Set("SIP/SIP-0xxxxxx38-00000001", "__MOHCLASS=default") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [0xxxxxx38@from-trunk:7] Set("SIP/SIP-0xxxxxx38-00000001", "__CALLINGPRES_SV=allowed_not_screened") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [0xxxxxx38@from-trunk:8] Set("SIP/SIP-0xxxxxx38-00000001", "CALLERPRES()=allowed_not_screened") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [0xxxxxx38@from-trunk:9] Goto("SIP/SIP-0xxxxxx38-00000001", "ivr-1,s,1") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Goto (ivr-1,s,1)
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [s@ivr-1:1] Set("SIP/SIP-0xxxxxx38-00000001", "TIMEOUT_LOOPCOUNT=0") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [s@ivr-1:2] Set("SIP/SIP-0xxxxxx38-00000001", "INVALID_LOOPCOUNT=0") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [s@ivr-1:3] Set("SIP/SIP-0xxxxxx38-00000001", "_IVR_CONTEXT_ivr-1=") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [s@ivr-1:4] Set("SIP/SIP-0xxxxxx38-00000001", "_IVR_CONTEXT=ivr-1") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [s@ivr-1:5] Set("SIP/SIP-0xxxxxx38-00000001", "__IVR_RETVM=") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [s@ivr-1:6] GotoIf("SIP/SIP-0xxxxxx38-00000001", "0?skip") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [s@ivr-1:7] Answer("SIP/SIP-0xxxxxx38-00000001", "") in new stack
[2015-05-23 12:02:25] VERBOSE[9440] pbx.c: -- Executing [s@ivr-1:8] Wait("SIP/SIP-0xxxxxx38-00000001", "1") in new stack
[2015-05-23 12:02:26] VERBOSE[9440] pbx.c: -- Executing [s@ivr-1:9] Set("SIP/SIP-0xxxxxx38-00000001", "IVR_MSG=custom/ivr_rec_ok") in new stack
[2015-05-23 12:02:26] VERBOSE[9440] pbx.c: -- Executing [s@ivr-1:10] Set("SIP/SIP-0xxxxxx38-00000001", "TIMEOUT(digit)=3") in new stack
[2015-05-23 12:02:26] VERBOSE[9440] func_timeout.c: -- Digit timeout set to 3.000
[2015-05-23 12:02:26] VERBOSE[9440] pbx.c: -- Executing [s@ivr-1:11] ExecIf("SIP/SIP-0xxxxxx38-00000001", "1?Background(custom/ivr_rec_ok)") in new stack
[2015-05-23 12:02:26] WARNING[9440] file.c: File custom/ivr_rec_ok does not exist in any format
[2015-05-23 12:02:26] WARNING[9440] file.c: Unable to open custom/ivr_rec_ok (format 0x8 (alaw)): No such file or directory
[2015-05-23 12:02:26] WARNING[9440] pbx.c: ast_streamfile failed on SIP/SIP-0xxxxxx38-00000001 for custom/ivr_rec_ok
[2015-05-23 12:02:26] VERBOSE[9440] pbx.c: -- Executing [s@ivr-1:12] WaitExten("SIP/SIP-0xxxxxx38-00000001", "5,") in new stack
[2015-05-23 12:02:31] VERBOSE[9440] pbx.c: == Spawn extension (ivr-1, s, 12) exited non-zero on 'SIP/SIP-0xxxxxx38-00000001'
[2015-05-23 12:02:31] VERBOSE[9440] pbx.c: -- Executing [h@ivr-1:1] Hangup("SIP/SIP-0xxxxxx38-00000001", "") in new stack
[2015-05-23 12:02:31] VERBOSE[9440] pbx.c: == Spawn extension (ivr-1, h, 1) exited non-zero on 'SIP/SIP-0xxxxxx38-00000001'

Overkill · May 23, 2015, 10:10am

The audio file should be 16-bit Mono PCM @ 8000Hz

Were these working before, or did they suddenly stop? If the latter, check Settings > Asterisk SIP Settings and make sure the correct codec(s) are checked.

Can you verify what “the correct location” is?

mannana93 · May 23, 2015, 10:17am

The file was working before and, as I said, if I put it on the moh it works.
For the test I checked all the codecs but I didn’t solve.
The file is in /var/lib/asterisk/sounds/custom

Overkill · May 23, 2015, 5:44pm

what is the file’s extension (case sensitive!)?

mannana93 · May 23, 2015, 6:24pm

The extension is .wav

Overkill · May 25, 2015, 4:07pm

Can you put the file up somewhere and I’ll check it out on my box? I’m pretty stumped at this point.

mannana93 · May 25, 2015, 4:49pm

dl.dropboxusercontent.com

mannana93 · May 27, 2015, 5:40pm

Do you have any new ideas ?

Overkill · May 28, 2015, 2:53pm

Sorry, still catching up from the holiday weekend.

I tried it on my PBX and it worked fine. The file you posted has a different name than “ivr_rec_ok”, but you seem smart enough that it’s probably not the issue

Have you tried uploading it as a totally different file/recording and using that?

mannana93 · May 28, 2015, 4:08pm

It’s the same file but with a different name that I used when uploading. I tried also this file on another FreePBX and it worked so there is something corrupted in my installation.

Overkill · May 28, 2015, 4:22pm

Can you remove it, add it with a different name, and use the newly uploaded copy?

mannana93 · May 28, 2015, 5:16pm

I tried also this but no change…