[asterisk-users] No audio playing back voicemail from odbc

Support mdiehl at diehlnet.com
Fri Aug 3 11:27:36 CDT 2012


Well, it gets even stranger....

I've installed version 10.2.1, instead of 10.7.1, and copied the configuration 
files from another identical server that is running 10.2.1 is working 
correctly.

I STILL can't get voicemail to play back.  I can hear the password prompts

Theses are, what I think to be, the relevant settings in voicemail.conf:

;minmessage=3
maxsilence=10
silencethreshold=128

When I set silencethresholdo either 500 or 64, I still didn't hear anything.  
(But I did hear several seconds of actual silence.  The .wav file contains 
nothing but silence.

So, fiddling with the silencethreshold in both directions, doesn't seem to 
change the symptoms.

Where else should I look?

TIA

Mike.

On Saturday 28 July 2012 6:45:15 pm Matthew Jordan wrote:
> ----- Original Message -----
> 
> > From: "Support" <mdiehl at diehlnet.com>
> > To: asterisk-users at lists.digium.com
> > Cc: "Matthew Jordan" <mjordan at digium.com>
> > Sent: Saturday, July 28, 2012 2:38:09 PM
> > Subject: Re: [asterisk-users] No audio playing back voicemail from odbc
> > 
> > 
> > CLI> core show translation paths slin
> > --- Translation paths SRC Codec "slin" sample rate 8000 ---
> > 
> >         slin       To g723      : No Translation Path
> >         slin       To gsm       : (slin)->(gsm)
> >         slin       To ulaw      : (slin)->(ulaw)
> > 
> > I'm using the ulaw audio codec and wav for storage, so this SHOULD
> > work....
> 
> If you're getting a duration of 0, I have to wonder if your
> silencethreshold is playing a factor here.  Asterisk may be treating the
> entire recording as silence.  What happens if you set the maxsilence to
> some valid integer value?  Does it end the voicemail messages while you're
> "leaving" it?  If so, that might indicate that it isn't detecting any
> sound.
> 
> If you set minduration/maxsilence and Asterisk starts killing recordings
> and not saving files, that will also tell you if Asterisk believes the
> recordings are mostly silence.
> 
> > I don't, but this configuration worked before I upgraded from 1.6.x
> > to 10.x.  I
> > should have mentioned that this was part of an upgrade, but it was
> > late, and I
> > was tired.
> > 
> > So, is there something I'm missing?
> 
> I'm not sure.  I'd be curious to see your voicemail.conf.
> 
> > > File storage is the only mechanism to have video voicemail (with
> > > audio)
> > > at this time.
> > 
> > Is there any interest in fixing this situation?  It doesn't seem like
> > it would
> > be too difficult.  I wouldn't mind helping if there is an effort
> > already
> > underway.
> 
> There has been some interest expressed from users, but no development
> plans have been put into place for this feature.
> 
> There are a couple of reasons for that: while it would be possible
> to have multiple formats stored in ODBC/IMAP backends, that doesn't solve
> all of the problems with associating an audio file with a video file.
> For example, some soft phones allow you to start the video media stream
> after the audio media stream has already begun.  This will work fine
> during the video call; however, if the video/audio is stored as a voicemail
> message, Asterisk has no way to associate the beginning of the video file
> with some arbitrary point in the audio file.  Hence, when the video
> message is played back, the video will be out of sync with the audio -
> both are played back starting at the same time, but the soft phone didn't
> start sending the video at the beginning of the audio.
> 
> The solution to this would be to store the audio/video as a single file
> in a media container (such as Matroska).  Not only does this solve the
> audio/video sync issue, but now you don't need to store more then a single
> file in a storage backend.  Unfortunately, this is an extremely non-trivial
> effort.
> 
> --
> Matthew Jordan
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
> 
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

-- 

Mike Diehl.



More information about the asterisk-users mailing list