Stefan Guenther
2007-Dec-03 18:01 UTC
[asterisk-users] Soundcard necessary on an asterisk server to get output of playback()??
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
********************************************
Lacy Moore
2007-Dec-03 18:54 UTC
[asterisk-users] Soundcard necessary on an asterisk server to get output of playback()??
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
Stefan Guenther
2007-Dec-03 21:59 UTC
[asterisk-users] Soundcard necessary on an asterisk server to get output of playback()??
Hi,
>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'm using ztdummy, and yes, I guess your're right - it seems to be a
timing problem, because I found the following messages in /var/log/messages:
Dec 3 22:51:36 asterisk kernel: [25713.830465] printk: 249 messages
suppressed.
Dec 3 22:51:36 asterisk kernel: [25713.830468] rtc: lost some
interrupts at 1024Hz.
BTW:
>name -a
Linux asterisk 2.6.22-14-386 #1 Sun Oct 14 22:36:54 GMT 2007 i686 GNU/Linux
But what does "lost some interrupts at 1024Hz" tell me? And if it is
related to my problem, how do I solve it?
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
********************************************
Stefan Guenther
2007-Dec-04 12:19 UTC
[asterisk-users] [SOLVED] Re: Soundcard necessary on an asterisk server to get output of playback()??
Hi,
I have found a solution for my problem or at least I can hear the output
of PLAYBACK() and the voicemail system.
Since some of you suggested a timing problem, I removed the ztdummy and
zaptel modules, but this had no effect.
For whatever reason I have to insert a WAIT(1) in front of every
application that returns an output. Well, now the context looks like
this and it works:
exten => 202,1,ANSWER()
exten => 202,2,WAIT(1)
exten => 202,3,PLAYBACK(tt-monkeys)
exten => 202,4,HANGUP()
Does anyone have an explanation why I need the WAIT() ?
Thanks for your help last night.
Stefan
BTW: As far as I can see, no one answered the question that I entered in
the subject line: Do I need a sound card in the server, does it have any
effect?
--
********************************************
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
********************************************
Bhrugu Mehta
2007-Dec-26 12:28 UTC
[asterisk-users] Soundcard necessary on an asterisk server to get output of playback()??
no , not at all, there is no need to install sound card in asteirsk system. I am using asterisk server without soundcard. so there may be antoher problem may in configurtion of zapata or other. cheers!!! Bhrugu mehta On Dec 3, 2007 11:31 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 >