[asterisk-bugs] [Asterisk 0017613]: [patch] After a blind transfer by the calling party the transferees peer cannot be dialed again within the same call
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue May 10 04:13:50 CDT 2011
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=17613
======================================================================
Reported By: ramonpeek
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 17613
Category: Channels/General
Reproducibility: always
Severity: minor
Priority: normal
Status: acknowledged
Asterisk Version: 1.4.33
JIRA: SWP-1833
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2010-07-09 05:38 CDT
Last Modified: 2011-05-10 04:13 CDT
======================================================================
Summary: [patch] After a blind transfer by the calling party
the transferees peer cannot be dialed again within the same call
Description:
After a blind transfer by the calling party the transferees peer cannot be
dialed again within the same call. This ONLY occurs when dialing through a
Local channel.
Asterisk will show this warning on the CLI>
[Jul 9 12:18:37] WARNING[27865]: app_dial.c:1296 dial_exec_full: Skipping
dialing interface 'SIP/401' again since it has already been dialed
NOTE: See steps to reproduce (in advanced view)
======================================================================
----------------------------------------------------------------------
(0134706) kaii (reporter) - 2011-05-10 04:13
https://issues.asterisk.org/view.php?id=17613#c134706
----------------------------------------------------------------------
UPDATE: did some re-tests of the scenario i described above with
1.8.4-rc3.
when manually forwarding the call (without answering it) to another
extension via SIP "302 Moved Temporarily", the forwarder is still not
removed from the list and the call is being hung up when trying to forward
it back the initial forwarder:
use case:
SIP/10 calls SIP/20
->> SIP/20 rings
SIP/20 manually forwards call to SIP/30
->> SIP/20 is now free, SIP/30 rings
SIP/30 manually forwards back to SIP/20
->> call is being hung up "Skipping dialing interface 'SIP/20' again"
will attach patch for 1.8.4-rc3 which removes the forwarder from the list
of dialed interfaces.
Issue History
Date Modified Username Field Change
======================================================================
2011-05-10 04:13 kaii Note Added: 0134706
======================================================================
More information about the asterisk-bugs
mailing list