[asterisk-bugs] [Asterisk 0017169]: no sound on Playback(<file>, noanswer)
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Apr 13 01:17:16 CDT 2010
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=17169
======================================================================
Reported By: leroi05
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 17169
Category: Applications/app_playback
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: 1.6.2.7-rc1
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2010-04-12 12:24 CDT
Last Modified: 2010-04-13 01:17 CDT
======================================================================
Summary: no sound on Playback(<file>,noanswer)
Description:
There is no sound if I try to playback a file without answering the channel
(sip, iax). Tried versoins 1.6.2.5, 1.6.2.6, 1.6.2.7-rc1, 1.6.1.19-rc1,
1.6.0.27-rc1.
On sip channel rtp debug gives nothing.
Reproduceable with minimal dialplan:
exten => _X.,1,Playback(demo-congrats,noanswer)
======================================================================
----------------------------------------------------------------------
(0120314) leroi05 (reporter) - 2010-04-13 01:17
https://issues.asterisk.org/view.php?id=17169#c120314
----------------------------------------------------------------------
I stated exactly on which versions I can reproduce this:
1.6.2.5, 1.6.2.6, 1.6.2.7-rc1, 1.6.1.19-rc1, 1.6.0.27-rc1
Sip clients (what doesn't matter in this issue):
ekiga 2.0.12
Snom 360 (fw version 8.2.25)
Siemens C470IP (fw version 022230000000 / 043.00)
Once again - all three work with production server just fine.
I can't understand what you don't undestand. What kind of debug
information do you need? Or I have to provide SIP/RTP trace for any
combination of sip client and asterisk version I've tried?
Issue History
Date Modified Username Field Change
======================================================================
2010-04-13 01:17 leroi05 Note Added: 0120314
======================================================================
More information about the asterisk-bugs
mailing list