Topic: Weird voicemail greeting playback issue after 3.1.1-147 update

[David Harper]

Hi all

I upgraded a site from an older 3.1 release of SAIL to the latest, 3.1.1-147.

After doing so, weird things are happening with voicemail. The main issue is that the call group I have set up to ring a few phones and then resolve to voicemail for one particular extension always plays the default voicemail message, rather than the extension's unavailable message.

Secondly, when one rings an extension directly, the voicemail plays the custom unavailable message PLUS the default message after that.

How can I modify this behaviour?

Thanks!
David

[SARK devs]

Hi David

Second point first.  Turn off voicemail instructions in globals.   
First point I'm not sure about - can you post a call trace from the Asterisk console please?

Cheers

S

[David Harper]

Okay, done the globals change. Thanks for that!

As for the main issue, here's the output when the call group goes direct to voicemail for extension 5000:

Code: [Select]

    -- Starting simple switch on 'DAHDI/3-1'
    -- Executing [s@from-pstn:1] Set("DAHDI/3-1", "chan=3-1") in new stack
    -- Executing [s@from-pstn:2] Set("DAHDI/3-1", "chan=3") in new stack
    -- Executing [s@from-pstn:3] Goto("DAHDI/3-1", "mainmenu,DAHDI3,1") in new stack
    -- Goto (mainmenu,DAHDI3,1)
    -- Sent into invalid extension 'DAHDI3' in context 'mainmenu' on DAHDI/3-1
    -- Executing [i@mainmenu:1] Goto("DAHDI/3-1", "extensions,5100,1") in new stack
    -- Goto (extensions,5100,1)
    -- Executing [5100@extensions:1] AGI("DAHDI/3-1", "sarkhpe,Alias,SIP/5000 SIP/5001 SIP/5008,5100,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Dial) Options: (SIP/5000&SIP/5001&SIP/5008,16,ctI)
  == Using SIP RTP CoS mark 5
    -- Called SIP/5000
  == Using SIP RTP CoS mark 5
    -- Called SIP/5001
  == Using SIP RTP CoS mark 5
    -- Called SIP/5008
    -- Connected line update to DAHDI/3-1 prevented.
    -- Connected line update to DAHDI/3-1 prevented.
    -- Connected line update to DAHDI/3-1 prevented.
    -- SIP/5001-00000dc2 is ringing
    -- SIP/5000-00000dc1 is ringing
    -- SIP/5008-00000dc3 is ringing
    -- Nobody picked up in 16000 ms
    -- <DAHDI/3-1>AGI Script sarkhpe completed, returning 0
    -- Executing [*5000@extensions:1] VoiceMail("DAHDI/3-1", "5000") in new stack
    -- <DAHDI/3-1> Playing 'vm-intro.alaw' (language 'en')
    -- <DAHDI/3-1> Playing 'beep.alaw' (language 'en')
    -- Recording the message
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/5000/tmp/QHRfrZ format: wav49, 0xb6d037c8

And here it is when it instead rings 5000 alone, which then times out to voicemail:

Code: [Select]

    -- Starting simple switch on 'DAHDI/3-1'
    -- Executing [s@from-pstn:1] Set("DAHDI/3-1", "chan=3-1") in new stack
    -- Executing [s@from-pstn:2] Set("DAHDI/3-1", "chan=3") in new stack
    -- Executing [s@from-pstn:3] Goto("DAHDI/3-1", "mainmenu,DAHDI3,1") in new stack
    -- Goto (mainmenu,DAHDI3,1)
    -- Sent into invalid extension 'DAHDI3' in context 'mainmenu' on DAHDI/3-1
    -- Executing [i@mainmenu:1] Goto("DAHDI/3-1", "extensions,5100,1") in new stack
    -- Goto (extensions,5100,1)
    -- Executing [5100@extensions:1] AGI("DAHDI/3-1", "sarkhpe,Alias,SIP/5000 SIP/5001 SIP/5008,5100,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Dial) Options: (SIP/5000&SIP/5001&SIP/5008,16,ctI)
  == Using SIP RTP CoS mark 5
    -- Called SIP/5000
  == Using SIP RTP CoS mark 5
    -- Called SIP/5001
  == Using SIP RTP CoS mark 5
    -- Called SIP/5008
    -- Connected line update to DAHDI/3-1 prevented.
    -- Connected line update to DAHDI/3-1 prevented.
    -- Connected line update to DAHDI/3-1 prevented.
    -- SIP/5008-00000dc6 is ringing
    -- SIP/5001-00000dc5 is ringing
    -- SIP/5000-00000dc4 is ringing
    -- Nobody picked up in 16000 ms
    -- <DAHDI/3-1>AGI Script sarkhpe completed, returning 0
    -- Executing [5000@extensions:1] AGI("DAHDI/3-1", "sarkhpe,InCall,,,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Dial) Options: (SIP/5000,20,kt)
  == Using SIP RTP CoS mark 5
    -- Called SIP/5000
    -- SIP/5000-00000dc7 is ringing
    -- Nobody picked up in 20000 ms
    -- AGI Script Executing Application: (Voicemail) Options: (5000,suu)
    -- <DAHDI/3-1> Playing '/var/spool/asterisk/voicemail/default/5000/unavail.gsm' (language 'en')
    -- <DAHDI/3-1> Playing 'beep.alaw' (language 'en')
    -- Recording the message
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/5000/tmp/nNJqWU format: wav49, 0xb6d097a0

Weird!

[SARK devs]

Ah, OK, I understand your question.

It is due to the way in which you are exercising voicemail with *{extension}, in the first example I believe.  I'm guessing you have a timeout target of *5000 for your callgroup.

In 3.x.x this will always play the stock vm-intro sound and ignore any unavailable message.   It's basically down to a design decision that was made at the time.  We believed (rightly or wrongly) that *{extension} would mostly be used manually to leave a message for a colleague or to transfer a call directly to a colleague's mailbox.  In such cases, it's annoying to listen to an unavailable message for someone you already know is unavailable. It was never really intended to be used to drive a general greeting mailbox such as you have done here, although in practice that does often happen.   

You can achieve what you want by either:-

timing out to the extension (5000) and waiting for it to ring out and drop naturally to voicemail

or:-

timing out to an IVR (with NO options specified).  This will play the unavailable message you want and then time out to *5000. 


Kind Regards

S

[David Harper]

Thanks Selintra!

timing out to the extension (5000) and waiting for it to ring out and drop naturally to voicemail

That's what we did temporarily - thanks for that!

timing out to an IVR (with NO options specified).  This will play the unavailable message you want and then time out to *5000

Can I copy the file /var/spool/asterisk/voicemail/default/5000/unavail.gsm to some other location and rename it so it shows up as a greeting for the purposes of the IVR, or does the site contact need to re-record their desired message?

Thanks
David

[David Harper]

Figured it out:

Code: [Select]

cd /var/spool/asterisk/voicemail/default/5000
sox unavail.WAV -r 8000 -c 1 -s -w /var/lib/asterisk/sounds/usergreeting0001.wav

Thanks - now works like a charm!

The only question I have is why did this work before with the earlier 3.x release? But that is a minor thing

[SARK devs]

The only question I have is why did this work before with the earlier 3.x release?

As I said, it was a design decision.  Many of our clients didn't want the unavail to play when transferring a call to a VMbox.   I can't remember when we changed it but it will be in the rpm changelog.

Best

S

[David Harper]

Thanks for the explanation!