Debian Squeeze + Asterisk + FreePBX

Hello everybody,

on a Debian Squeeze I installed Asterisk 1.6 e FreePBX 2.8, I configured extensions and trunks and it works fine.

Now, I would like to set up an IVR so that when someone calls he listen to a message with instructions, then he can dial an internal extension or wait and go to a queue.

I can’t make two things work:

  1. default messages and messages I uploaded seem to be ignored (format should be correct), even if into CLI they seem they are played

  2. in IVR direct dial is enabled for all extensions but it doesn’t work: into CLI I can’t see anything when I digit an extension from outside. The strange thing is that it seems no dial tone reaches Asterisk…

Any idea?

Best regards,
Matteo

I did, but I can’t understand the problem… Whan I call from outside I get:

=======================================================
[2010-12-16 16:01:21] VERBOSE[11675] netsock.c: == Using SIP RTP TOS bits 184
[2010-12-16 16:01:21] VERBOSE[11675] netsock.c: == Using SIP RTP CoS mark 5
[2010-12-16 16:01:21] VERBOSE[11720] pbx.c: – Executing [[email protected] smart:1] Set(“SIP/Messagent smart-00000000”, “GROUP()=OUT_1”) in new stack
[2010-12-16 16:01:21] VERBOSE[11720] pbx.c: – Executing [[email protected] smart:2] Goto(“SIP/Messagent smart-00000000”, “from-trunk,01731996,1”) in new stack
[2010-12-16 16:01:21] VERBOSE[11720] pbx.c: – Goto (from-trunk,01731996,1)
[2010-12-16 16:01:21] VERBOSE[11720] pbx.c: – Executing [[email protected]:1] NoOp(“SIP/Messagent smart-00000000”, “Catch-All DID Match - Found 01731996 - You probably want a DID for this.”) in new stack
[2010-12-16 16:01:21] VERBOSE[11720] pbx.c: – Executing [[email protected]:2] Goto(“SIP/Messagent smart-00000000”, “ext-did,s,1”) in new stack
[2010-12-16 16:01:21] VERBOSE[11720] pbx.c: – Goto (ext-did,s,1)
[2010-12-16 16:01:21] VERBOSE[11720] pbx.c: – Executing [[email protected]:1] Set(“SIP/Messagent smart-00000000”, “__FROM_DID=s”) in new stack
[2010-12-16 16:01:21] VERBOSE[11720] pbx.c: – Executing [[email protected]:2] Set(“SIP/Messagent smart-00000000”, “CHANNEL(language)=it”) in new stack
[2010-12-16 16:01:21] VERBOSE[11720] pbx.c: – Executing [[email protected]:3] ExecIf(“SIP/Messagent smart-00000000”, “0 ?Set(CALLERID(name)=+3939335340)”) in new stack
[2010-12-16 16:01:21] VERBOSE[11720] pbx.c: – Executing [[email protected]:4] SetMusicOnHold(“SIP/Messagent smart-00000000”, “none”) in new stack
[2010-12-16 16:01:21] WARNING[11720] res_musiconhold.c: SetMusicOnHold application is deprecated and will be removed. Use Set(CHANNEL(musicclass)=…) instead
[2010-12-16 16:01:21] VERBOSE[11720] pbx.c: – Executing [[email protected]:5] Set(“SIP/Messagent smart-00000000”, “__MOHCLASS=none”) in new stack
[2010-12-16 16:01:21] VERBOSE[11720] pbx.c: – Executing [[email protected]:6] Ringing(“SIP/Messagent smart-00000000”, “”) in new stack
[2010-12-16 16:01:21] VERBOSE[11720] pbx.c: – Executing [[email protected]:7] Wait(“SIP/Messagent smart-00000000”, “2”) in new stack
[2010-12-16 16:01:23] VERBOSE[11720] pbx.c: – Executing [[email protected]:8] Set(“SIP/Messagent smart-00000000”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
[2010-12-16 16:01:23] VERBOSE[11720] pbx.c: – Executing [[email protected]:9] Set(“SIP/Messagent smart-00000000”, “CALLERPRES()=allowed_not_screened”) in new stack
[2010-12-16 16:01:23] VERBOSE[11720] pbx.c: – Executing [[email protected]:10] Goto(“SIP/Messagent smart-00000000”, “ivr-3,s,1”) in new stack
[2010-12-16 16:01:23] VERBOSE[11720] pbx.c: – Goto (ivr-3,s,1)

… till there it seems ok…

=======================================================
[2010-12-16 16:01:23] VERBOSE[11720] pbx.c: – Executing [[email protected]:1] Set(“SIP/Messagent smart-00000000”, “MSG=custom/welcomehere”) in new stack
[2010-12-16 16:01:23] VERBOSE[11720] pbx.c: – Executing [[email protected]:2] Set(“SIP/Messagent smart-00000000”, “LOOPCOUNT=0”) in new stack
[2010-12-16 16:01:23] VERBOSE[11720] pbx.c: – Executing [[email protected]:3] Set(“SIP/Messagent smart-00000000”, “__DIR-CONTEXT=”) in new stack
[2010-12-16 16:01:23] VERBOSE[11720] pbx.c: – Executing [[email protected]:4] Set(“SIP/Messagent smart-00000000”, “_IVR_CONTEXT_ivr-3=”) in new stack
[2010-12-16 16:01:23] VERBOSE[11720] pbx.c: – Executing [[email protected]:5] Set(“SIP/Messagent smart-00000000”, “_IVR_CONTEXT=ivr-3”) in new stack
[2010-12-16 16:01:23] VERBOSE[11720] pbx.c: – Executing [[email protected]:6] GotoIf(“SIP/Messagent smart-00000000”, “0?begin”) in new stack
[2010-12-16 16:01:23] VERBOSE[11720] pbx.c: – Executing [[email protected]:7] Answer(“SIP/Messagent smart-00000000”, “”) in new stack
[2010-12-16 16:01:23] VERBOSE[11720] pbx.c: – Executing [[email protected]:8] Wait(“SIP/Messagent smart-00000000”, “1”) in new stack
[2010-12-16 16:01:24] VERBOSE[11720] pbx.c: – Executing [[email protected]:9] Set(“SIP/Messagent smart-00000000”, “TIMEOUT(digit)=3”) in new stack
[2010-12-16 16:01:24] VERBOSE[11720] func_timeout.c: – Digit timeout set to 3.000
[2010-12-16 16:01:24] VERBOSE[11720] pbx.c: – Executing [[email protected]:10] Set(“SIP/Messagent smart-00000000”, “TIMEOUT(response)=10”) in new stack
[2010-12-16 16:01:24] VERBOSE[11720] func_timeout.c: – Response timeout set to 10.000
[2010-12-16 16:01:24] VERBOSE[11720] pbx.c: – Executing [[email protected]:11] Set(“SIP/Messagent smart-00000000”, “__IVR_RETVM=”) in new stack
[2010-12-16 16:01:24] VERBOSE[11720] pbx.c: – Executing [[email protected]:12] ExecIf(“SIP/Messagent smart-00000000”, “1?Background(custom/welcomehere)”) in new stack
[2010-12-16 16:01:24] WARNING[11720] format_wav.c: Unexpected frequency 44100
[2010-12-16 16:01:24] WARNING[11720] file.c: Unable to open format wav
[2010-12-16 16:01:24] WARNING[11720] file.c: Unable to open custom/benvenutiinors (format 0x8 (alaw)): No such file or directory
[2010-12-16 16:01:24] WARNING[11720] pbx.c: ast_streamfile failed on SIP/Messagent smart-00000000 for custom/benvenutiinors

Very strange: mediainfo welcomehere.wav gives:

General
Complete name : welcomehere.wav
Format : Wave
File size : 84.3 KiB
Duration : 5s 394ms
Overall bit rate : 128 Kbps

Audio
ID : 0
Format : PCM
Codec ID : 1
Codec ID/Hint : Microsoft
Duration : 5s 394ms
Bit rate : 128 Kbps
Channel(s) : 1 channel
Sampling rate : 8 000 Hz
Bit depth : 16 bits
Stream size : 84.2 KiB (100%)

Here I shoud insert an extension, but nothing happens:

=======================================================
[2010-12-16 16:01:24] VERBOSE[11720] pbx.c: – Executing [[email protected]:13] WaitExten(“SIP/Messagent smart-00000000”, “,”) in new stack
[2010-12-16 16:01:34] VERBOSE[11720] pbx.c: – Timeout on SIP/Messagent smart-00000000, going to ‘t’
[2010-12-16 16:01:34] WARNING[11720] func_db.c: DB_DELETE requires an argument, DB_DELETE(/)
[2010-12-16 16:01:34] VERBOSE[11720] pbx.c: – Executing [[email protected]:1] NoOp(“SIP/Messagent smart-00000000”, "Deleting: ") in new stack
[2010-12-16 16:01:34] VERBOSE[11720] pbx.c: – Executing [[email protected]:2] Set(“SIP/Messagent smart-00000000”, “__NODEST=”) in new stack
[2010-12-16 16:01:34] VERBOSE[11720] pbx.c: – Executing [[email protected]:3] Goto(“SIP/Messagent smart-00000000”, “ext-queues,500,1”) in new stack

Then control passes to queues:

=======================================================
[2010-12-16 16:01:34] VERBOSE[11720] pbx.c: – Goto (ext-queues,500,1)
[2010-12-16 16:01:34] VERBOSE[11720] pbx.c: – Executing [[email protected]:1] Macro(“SIP/Messagent smart-00000000”, “user-callerid,”) in new stack
[2010-12-16 16:01:34] VERBOSE[11720] pbx.c: – Executing [[email protected]:1] Set(“SIP/Messagent smart-00000000”, “AMPUSER=+3939335340970”) in new stack
[…]

Perhaps you should tail -f /var/log/asterisk/full and see what is really happening, the cli doesn’t tell much sometimes.

… even if I do all locally: I created a queue that falls back to the IVR. When I call the queue, it passes correctly control to IVR, but I still can’t ear any announce, nor I can dial any extensions (actually I can, but they seem to be ignored).

Any help?

Best regards,
Matteo

I’ve faced the same issues with Debian Squeeze and Ubuntu Lucid using asterisk 1.8 packages provided by asterisk.org.

I’ve tried converting the files into gsm, sln, ulaw but they just don’t play.

To confirm that these were not user errors, I installed CentOS 5.6 + asterisk 1.8 and freepbx 2.9. All files played fine.

Under Ubuntu there is a link /usr/share/asterisk/sounds/custom -> …/…/…/local/share/asterisk/sounds

Changing the link to point to /var/lib/asterisk/sounds/custom fixed my issue.