FreePBX User Portal

Hello!

I am using FreePBX 2.5 throught installation of AsteriskNOW 1.5. I leave messages to one extension but I cannot see any recordings when I log in into the FreePBX User Portal. Do you know why? Must I configure any param?

Regards

Any idea?

You need to provide some info for us to go on.

You need to read (or post a section of) the logs when a VM is being left.

If you do you know how to do these things then that is where you
should start out, learning How to trouble shoot your install.

The asterisk log file will give a clue

Calling to a extension (69) Voicemail I get:

-- <SIP/10-082ad460> Playing 'vm-theperson' (language 'es')
-- <SIP/10-082ad460> Playing 'digits/6' (language 'es')
-- <SIP/10-082ad460> Playing 'digits/9' (language 'es')
-- <SIP/10-082ad460> Playing 'vm-isunavail' (language 'es')
-- <SIP/10-082ad460> Playing 'vm-intro' (language 'es')
-- <SIP/10-082ad460> Playing 'beep' (language 'es')
-- Recording the message
-- x=0, open writing:  /var/spool/asterisk/voicemail/default/69/tmp/n9B6Ww format: wav49, 0x8266dc0
-- x=1, open writing:  /var/spool/asterisk/voicemail/default/69/tmp/n9B6Ww format: wav, 0x81f9508
-- User hung up
-- Recording was 2 seconds long but needs to be at least 3 - abandoning

== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on ‘SIP/10-082ad460’ in macro ‘vm’
== Spawn extension (macro-exten-vm, s, 18) exited non-zero on ‘SIP/10-082ad460’ in macro ‘exten-vm’
== Spawn extension (from-internal, 69, 1) exited non-zero on ‘SIP/10-082ad460’
– Executing [[email protected]:1] Macro(“SIP/10-082ad460”, “hangupcall”) in new stack
– Executing [[email protected]:1] ResetCDR(“SIP/10-082ad460”, “w”) in new stack
– Executing [[email protected]:2] NoCDR(“SIP/10-082ad460”, “”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/10-082ad460”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [[email protected]:6] GotoIf(“SIP/10-082ad460”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/10-082ad460”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [[email protected]:11] Hangup(“SIP/10-082ad460”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/10-082ad460’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/10-082ad460’

I have recording file of message left but I don’t see any voicemail messages through FreePBX User Portal. Must I configure or modify path to recordings?

Regards

The line from your log explains what happened:

– Recording was 2 seconds long but needs to be at least 3 - abandoning

Leave longer messages and see if they get recorded.

I’ve enabled call recording for all calls to one extension. I can see recording files into /var/spool/asterisk/monitor but no recordings appear into FreePBX recording panel… I have also recording files into voicemail recording folder and they don’t appear via FreePBX web panel… Anyone knows why?

Regards

Please verify that the recording actually exists on your system (as it can’t show something that is not there), then check its permissions and ownership which should be asterisk:asterisk. If the files permissions are not set correctly it can exist in the system but the ARI will not see it as it will not have the correct rights.

Recording files (voicemail and monitored calls) exits as I can see them into their folders, /var/spool/asterisk/monitor and voicemail folder. All files and folder has asterisk as owner and group so I think is not a rights problem… Any idea?

is apache running as the user asterisk or as the user apache?

it runs as ‘apache’

I’ve just changed httpd owner to ‘asterisk’ and now I can see calls and Monitor link but when I try to play any of them a blank window opens but no sound plays…

You didn’t say which version of AsteriskNow you are using 1.5 Beta 1 or Beta 2. If you are using Beta 1 I’d take a look at Beta 2 first. Next off this sounds like a setup configuration issue with AsteriskNow and you might be better off over in that forum then here (or use the #asterisknow IRC channel).

Posting a solution here would be nice for those that have this issue at a later date.

Make sure you have the latest module. FreePBX ARI Framework is currently 2.5.2.1.

Try the download link instead of the play link for the voicemail. If you can download the recordings through your browser, that means ARI is working fine, but that your browser does not have correct plugins and/or configuration to play the recordings as streaming media. This is a very common problem for Firefox on most Linux distros. I spent a good part of an afternoon fixing this issue in CentOS 5.2. Also, Windows often doesn’t handle GSM well until Quicktime is installed.

Also, if you use the Call Me link for the voicemail playback, make sure you have a valid Call Me Number defined in your ARI Settings page before initiating the Call Me.