[asterisk-bugs] [Asterisk 0017134]: Asterisk 1.4.30 crashes on transers with the patch around CONNECTEDLINE (https://issues.asterisk.org/view.php?id=8824#118065)
Asterisk Bug Tracker
noreply at bugs.digium.com
Sun Apr 4 22:24:39 CDT 2010
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=17134
======================================================================
Reported By: leff
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 17134
Category: Functions/NewFeature
Reproducibility: always
Severity: crash
Priority: normal
Status: new
Asterisk Version: 1.4.30
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2010-04-03 14:01 CDT
Last Modified: 2010-04-04 22:24 CDT
======================================================================
Summary: Asterisk 1.4.30 crashes on transers with the patch
around CONNECTEDLINE (https://issues.asterisk.org/view.php?id=8824#118065)
Description:
The patch around CONNECTEDLINE function
(https://issues.asterisk.org/view.php?id=8824#118065) is applied to
Asterisk 1.4.30
Everything is fine but Asterisk crashes right after consultative transfer
(at least SIP REFER).
The example:
DAHDI (or IAX2 at least) 1 -> SIP 1 (Set(CONNECTEDLINE...))
SIP 1 answers DAHDI 1.
SIP1 -(over second line)-> SIP 2 (Set(CONNECTEDLINE...))
SIP 2 answers SIP 1
SIP 1 transfers the call (SIP REFER)
Asterisk crashes.
Is there workaround?
======================================================================
----------------------------------------------------------------------
(0120131) licedey (reporter) - 2010-04-04 22:24
https://issues.asterisk.org/view.php?id=17134#c120131
----------------------------------------------------------------------
I also tested this patch and found another two issue:
1) Connectedline feature doesn't work with IVR, or with answer()
application.
2) When we transfer inbound sip call, the call is hanged after 10~20
seconds. After checking sip dump, I found that after applying the patch sip
starts sending double invite messages.
Issue History
Date Modified Username Field Change
======================================================================
2010-04-04 22:24 licedey Note Added: 0120131
======================================================================
More information about the asterisk-bugs
mailing list