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 >