Summary:ASTERISK-15330: [regression] Record application hangs up after exactly 30 seconds, with or without silence or duration specified
Reporter:John S. (johnakabean)Labels:
Date Opened:2009-12-17 14:59:46.000-0600Date Closed:2011-06-07 14:01:04
Versions:Frequency of
Environment:Attachments:( 0) 0116602.txt
Description:The Record application hangs up after exactly 30 seconds, with or without silence or duration specified; I reverted to and the problem disappeared. My first instinct was session timers were enabled on the softphone I experienced the problem with.

Ehh, log entry says it has nothing to do with the record application BUT it only happens ON the record application.

[Dec 17 15:30:04] DEBUG[29018] res_musiconhold.c: Read 490 bytes of audio while expecting 640
[Dec 17 15:30:37] DEBUG[29071] app_record.c: Got hangup
Comments:By: Leif Madsen (lmadsen) 2009-12-18 07:31:19.000-0600

Would you be able to reproduce this and provide a backtrace? That may be useful for a developer to help narrow down why this might be happening.

If you could also provide the simple dialplan that reproduces this along with your sip.conf file, that may also provide useful in reproducing the problem.


By: Fernando Lujan (flujan) 2009-12-18 19:17:16.000-0600

Same here.

By: Fernando Lujan (flujan) 2009-12-28 09:30:50.000-0600

I update my setup to version 1.4.28 and I am still having this issue on my box. Which additional information do you require?

By: Leif Madsen (lmadsen) 2010-01-04 13:57:04.000-0600

Please see this note:  https://issues.asterisk.org/view.php?id=16462#115419

By: Fernando Lujan (flujan) 2010-01-13 10:46:48.000-0600

I just attached a file with the information from the 1.4.29-rc1. I am not seeing the app_record debug messages, just the frame not received from the extension.

Here it goes.

By: Leif Madsen (lmadsen) 2010-01-13 12:26:13.000-0600

Per IRC, the revision this was noticed in is 221086

By: Leif Madsen (lmadsen) 2010-01-13 12:26:40.000-0600

If the reporter could try a revision prior to that and report back if they are still having an issue or not, that would be advisable. Thanks!

By: Fernando Lujan (flujan) 2010-01-15 07:24:11.000-0600

lmaden, I tried with version 1.4.29-rc1. The workaround is to set the "constantssrc=true" in the sip.conf file.

This works for the manual calls, but I got a strange behavior with calls dialed using the originate AMI comamnd. I will wait to the final version of 1.4.29 and report a new issue if the problem with the originate not be fixed.


By: Leif Madsen (lmadsen) 2010-02-03 14:22:52.000-0600

Closed per the reporter as the work around was found, and they will report a new issue if the other issue found is not resolved in latest releases.