[asterisk-users] Soundcard necessary on an asterisk server to get output of playback()??

[Lacy Moore]

My quick guess would be that it's a timing issue.  You didn't mention
whether you are using a Zaptel device or ztdummy.

I know this sounds like I'm being a smart***, but I'm not...  try this...
rub the mouthpiece of the file while the sound file is playing and see if
you hear any of the file.  If so, I would definitely say you have a timing
issue.

On Dec 3, 2007 12:01 PM, Stefan Guenther <asterisk01 at in-put.de> wrote:

> Hi,
>
> I' still fighting the problem, that I can talk from one SIP phone to
> another, but I can't hear the output of the playback or similar
> applications:
>
>         exten => 202,1,ANSWER()
>         exten => 202,2,PLAYBACK(tt-monkeys)
>         exten => 202,3,HANGUP()
>
> When I dial 202, asterisk show the following on the cli:
>
> -- Executing [202 at local:1] Answer("SIP/user1-0827ebe8", "") in new stack
> -- Executing [202 at local:2] Playback("SIP/user1-0827ebe8", "tt-monkeys")
> in new stack
> -- <SIP/user1-0827ebe8> Playing 'tt-monkeys' (language 'de')
>
> Yes, the file tt-monkeys exist in /var/lib/asterisk/sounds and the
> subdirectory de.
>
> No, there is no error message even if turn on debugging. :-(
>
> Besides this strange behaviour, I was wondering whether the asterisk
> server needs an soundcard to send the output of e.g. the playback
> application to the phone.
>
> BTW, this is asterisk 1.4.13
>
> I would be really happy, if someone has an idea how to solve this problem.
>
> Thanks in advance,
>
> 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
>



-- 
Lacy Moore
Somewhere I wish I wasn't
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071203/33e6a835/attachment.htm