[asterisk-users] Soundcard necessary on an asterisk server to get output of playback()??
Jason Parker
jparker at digium.com
Mon Dec 3 13:59:50 CST 2007
I think Lacy means "rub the mouthpiece of the phone" - to make sound (blowing
into it should yield the same result)
Lacy Moore wrote:
> 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
> <mailto: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 <http://www.in-put.de/>
> ********************************************
> Schulungen Installationen
> Beratung Support
> Voice-over-IP-Loesungen
> ********************************************
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> <http://www.api-digital.com--/>
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
> <http://lists.digium.com/mailman/listinfo/asterisk-users>
>
>
>
>
> --
> Lacy Moore
> Somewhere I wish I wasn't
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> --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
--
Jason Parker
Digium
More information about the asterisk-users
mailing list