asterisk users - Nov 2007 - No sound from playback and voicemail

Hello,

I have a strange situation:

I can talk to other SIP phones and via ISDN to the outside, but I don't hear
playbacks or the voicemail messages.
Asterisk show in the cli, that the corresponding files are played, but I hear
nothing at all.

Here is as simple example:

[monkeys]
??? exten => 99,1,ANSWER()
??? exten => 99,2,PLAYBACK(tt-monkeys)
??? exten => 99,3,HANGUP()

The phone has access to this context, and the file exists, all codecs are
allowed.
I have tried to load either chan_alsa.so or chan_oss.so but it doesn't
change
anything.

Does anyone have an idea what could be wrong? This is not the first Asterisk
system that I set up, but I never had a problem like this.

Asterisk is version 1.4.13

Thanks for your help,

Stefan

-- 

********************************************
in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Geschaeftsfuehrer
Moltkestrasse 49? ? ?D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de
********************************************
? ? ?Schulungen? Installationen? 
? ? ? ? ?Beratung? ?Support
? ? ? Voice-over-IP-Loesungen
********************************************

On Mon, 2007-11-12 at 15:46 +0100, Stefan Guenther
wrote:
> Hello,
> 
> I have a strange situation:
> 
> I can talk to other SIP phones and via ISDN to the outside, but I don't
hear
> playbacks or the voicemail messages.
> Asterisk show in the cli, that the corresponding files are played, but I
hear
> nothing at all.
> 
	I once had this exact problem on a new installation of Asterisk and it
was because one of the cards was not properly seated in the motherboard.
Once I pushed the card all the way in I could hear voicemail and the
IVR.

-- 
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez Prats
Director de Tecnolog?a
+52-55-91169161 ext 2001
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 189 bytes
Desc: This is a digitally signed message part
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20071112/41485581/attachment.pgp

Stefan Guenther wrote:
> Hello,
> 
> I have a strange situation:
> 
> I can talk to other SIP phones and via ISDN to the outside, but I don't
hear
> playbacks or the voicemail messages.
> Asterisk show in the cli, that the corresponding files are played, but I
hear
> nothing at all.
> 
> Here is as simple example:
> 
> [monkeys]
>     exten => 99,1,ANSWER()
>     exten => 99,2,PLAYBACK(tt-monkeys)
>     exten => 99,3,HANGUP()
> 
> The phone has access to this context, and the file exists, all codecs are
> allowed.
> I have tried to load either chan_alsa.so or chan_oss.so but it doesn't
change
> anything.
> 
> Does anyone have an idea what could be wrong? This is not the first
Asterisk
> system that I set up, but I never had a problem like this.
> 
> Asterisk is version 1.4.13
Do you have zaptel loaded? Could this be related?

http://lists.digium.com/pipermail/asterisk-users/2007-November/199784.html

If you do - you can try downgrading zaptel version.

Regards,
Atis

This will also happen if there is a zap card installed and  
unconfigured in zaptel.conf & zapata.conf.


Forrest Beck
jonforrest.beck at gmail.com
www.shift8.biz
dCAP


On Nov 12, 2007, at 9:46 AM, Stefan Guenther wrote:
> Hello,
>
> I have a strange situation:
>
> I can talk to other SIP phones and via ISDN to the outside, but I  
> don't hear
> playbacks or the voicemail messages.
> Asterisk show in the cli, that the corresponding files are played,  
> but I hear
> nothing at all.
>
> Here is as simple example:
>
> [monkeys]
>     exten => 99,1,ANSWER()
>     exten => 99,2,PLAYBACK(tt-monkeys)
>     exten => 99,3,HANGUP()
>
> The phone has access to this context, and the file exists, all  
> codecs are
> allowed.
> I have tried to load either chan_alsa.so or chan_oss.so but it  
> doesn't change
> anything.
>
> Does anyone have an idea what could be wrong? This is not the first  
> Asterisk
> system that I set up, but I never had a problem like this.
>
> Asterisk is version 1.4.13
>
> Thanks for your help,
>
> Stefan
>
> -- 
>
> ********************************************
> in-put GbR - Das Linux-Systemhaus
> Stefan-Michael Guenther
> Geschaeftsfuehrer
> Moltkestrasse 49     D-76133 Karlsruhe
> Tel./Fax : +49 (0)721 / 83044 - 98/93
> http://www.in-put.de
> ********************************************
>      Schulungen  Installationen
>          Beratung   Support
>       Voice-over-IP-Loesungen
> ********************************************
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20071112/5d94b8ad/attachment.htm