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Summary:ASTERISK-12954: Cannot record sounds in WAV and WAV49 longer then 5-8 seconds
Reporter:Tech Admin (tech_admin)Labels:
Date Opened:2008-10-23 03:22:25Date Closed:2011-06-07 14:02:53
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Applications/app_record
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) 2008-112-19-greet-1.tmp.WAV
( 1) 2008-12-17-greet-1.tmp.WAV
Description:Hello,
We leave a voicemail for 30 seconds, then hangup, the audio file records only the first 5 to 9 seconds. we have tried the same with the record command and the result is the same.
this is valid for recording in WAV and WAV49.

we tried in different codecs for the call g729, and g711 doesn't work either
recording in GSM work fine.
the # when in WAV or WAV49 doesn't work.
We are now doing the recording on an other server with asterisk 1.4.21.2 and it works fine.
TA

****** ADDITIONAL INFORMATION ******

Configurations
Server : HP DL320 G5
Kernel : rhel5 2.6.18-92.1.13.el5
asterisk : 1.6.0
TC400B board
DAHDI 2.0
Comments:By: Tech Admin (tech_admin) 2008-10-23 03:37:35

On the 1.4.21 server recordign worh in WAV and not WAV49
Refer to http://bugs.digium.com/view.php?id=13588

By: Terry Wilson (twilson) 2008-12-16 14:43:46.000-0600

I just recorded a 30 second wav49 voicemail and played it back with absolutely no problems on Asterisk SVN-branch-1.6.0-r164804 built by terry @ cesium on a x86_64 running Linux on 2008-12-11 17:17:01 UTC.  I suggest trying to check out the 1.6.0 branch from SVN and seeing if that fixes your issue.

By: Terry Wilson (twilson) 2008-12-16 15:20:30.000-0600

I have no TC400B, so if someone who does can test this, that'd be great.

By: Tech Admin (tech_admin) 2008-12-16 15:56:03.000-0600

We are comipling one box with the card right now,give us some time, we'll post back the answers

By: Tech Admin (tech_admin) 2008-12-16 17:00:57.000-0600

We have tested on
Asterisk SVN-branch-1.6.0-r164878M built on a i686 running Linux on 2008-12-16 21:53:38 UTC
Conclusion
Voicemial message still chopped and: we can see the file growing in size during recording. After recording is finished we just hear the only the first 2 -3 seconds.
file is attached
also we have tried g723@6.3, robotik chopped sound.
Any ideas?

By: Terry Wilson (twilson) 2008-12-16 17:05:09.000-0600

tech_admin: And if you try it on a machine with no transcoder card?  Do you still get the issue?  I'm just trying to figure out why it works for me, and doesn't for you.

By: Tech Admin (tech_admin) 2008-12-16 17:19:11.000-0600

I can give you ssh access to the box in question throught a terminal client.
want to try it?

By: Shaun Ruffell (sruffell) 2008-12-17 11:41:17.000-0600

tech_admin:  Could you try with the http://svn.digium.com/svn/asterisk/team/sruffell/asterisk-trunk-transcoder or http://svn.digium.com/svn/asterisk/team/sruffell/asterisk-1.4-transcoder branches?

I know running "file convert" doesn't work with asterisk-1.4 or asterisk-trunk and codec_dahdi (you should get a file of size 0) because codec_dahdi, as it is now, just submits frames to the transcoder asynchronously from reading them out.  File convert expects a 1-to-1 correlation between packets in and packets out and believes it is done the first time it looks for a packet on the way out and one isn't there (because the hardware is still transcoding it)

The branches actually will wait around 50ms for a packet to finish transcoding, so the file convert should work there (but this increases probability of "avoiding deadlock" messages since the lock is held by a thread that is sleeping).

The branches also should fix the problem with robotik audio with the G723, but I haven't tried "file convert" with a G723 file.



By: Tech Admin (tech_admin) 2008-12-17 16:09:08.000-0600

ok, here are results for http://svn.digium.com/svn/asterisk/team/sruffell/asterisk-trunk-transcoder
G723: audio was good and usable, but: I called using g723 from a linksys phone to pstn. the linksys phone heard very well and in real time the pstn. But the pstn side heard the words said in the linksys phone 90 seconds later.
Voicemail still not working, hear attached file, audio is ok but chopped after a 3 seconds.

Sound quality is ok now.

TA

By: Shaun Ruffell (sruffell) 2008-12-18 16:55:50.000-0600

Hmm....ok, then the next step is for me to setup up this scenario in my lab here.  Thanks for the feedback.

By: Tech Admin (tech_admin) 2008-12-18 20:53:05.000-0600

We have as well setup a fedora with no accelerator board on it and will do a few tests too and post results.

By: Leif Madsen (lmadsen) 2009-01-20 13:16:03.000-0600

Any results here to report?

By: Tech Admin (tech_admin) 2009-01-20 16:15:32.000-0600

Yes, as mentioned to Trey in email exchange, we have come to conclusion that the recording of file is an OS issue ( the way files are written,...)and not astersik, since with fedora all works. I guess th enst step woudl be to post a but with RHEL 5. but what should be said? any ideas?

By: Tech Admin (tech_admin) 2009-01-22 04:44:45.000-0600

We probably have identified the culprit:
We have made a clean reinstall on a server containing the tcb400 card with RHEL 5.3 and latest asterisk + dadhi.
we run some test for voice mail recording.
When recording message on the server all is fine and works as it should, BUT,
When recording messages on an NFS share, we get corrupted files.
We strongly suspect NFS to be the problem. Can anyone repeat the test to confirm it? We are making more test and will come up with more results later.

By: Leif Madsen (lmadsen) 2009-01-22 09:59:36.000-0600

Thanks for the updates tech_admin!

It is generally acknowledged that writing directly to an NFS from any part of Asterisk is going to cause problems, whether that be with locking, or file corruption as you have pointed out.

So based on that, I'm not surprised that NFS is really your issue, and that if there is any way you can batch things out so that Asterisk is more writing to the local drive, and then using an external script to move things over to the NFS partition (so Asterisk doesn't have to access it directly), then you'll probably find you won't have the issues that you are running into.

Since this appears to no longer be an Asterisk issue, and that you have narrowed down to where the issues lies in your system, would it be fair for me to close this issue?

Thanks!
Leif.

By: Terry Wilson (twilson) 2009-01-22 10:05:09.000-0600

Weird, we used NFS at Nuvio for voicemail w/ no problems at all.

By: Tech Admin (tech_admin) 2009-01-22 10:06:40.000-0600

Yes you can, thanks for your help!

By: Tech Admin (tech_admin) 2009-01-22 10:07:28.000-0600

You an also close
http://bugs.digium.com/view.php?id=13588
Thanks

By: Leif Madsen (lmadsen) 2009-01-22 12:17:12.000-0600

Closed at the request of the reporter. Thanks!