[asterisk-bugs] [Asterisk 0012713]: [patch] SIP Protocol Violation when REFER rejected in sip_transfer (Cisco CCM, post answer), and Transfer application misclaims
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Dec 1 09:41:04 CST 2009
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=12713
======================================================================
Reported By: davidw
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 12713
Category: Channels/chan_sip/Transfers
Reproducibility: always
Severity: minor
Priority: normal
Status: acknowledged
Asterisk Version: 1.4.20
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2008-05-23 12:59 CDT
Last Modified: 2009-12-01 09:41 CST
======================================================================
Summary: [patch] SIP Protocol Violation when REFER rejected
in sip_transfer (Cisco CCM, post answer), and Transfer application misclaims
Description:
A call from Cisco CCM was answered and then transferred back to a different
extension on the Cisco.
Asterisk first reported a successful TRANSFERSTATUS, then tried to use a
REFER method. This was rejected as unsupported. Asterisk responded with
BYE, which the Cisco accepted. However it continued to try to send REFERs,
which the Cisco, of course, rejected, because the session reference had
been deleted by the BYE!
======================================================================
Relationships ID Summary
----------------------------------------------------------------------
related to 0010052 NOTIFY race condition when state change...
related to 0011848 Incorrect dialog matching and requests ...
======================================================================
----------------------------------------------------------------------
(0114447) davidw (reporter) - 2009-12-01 09:41
https://issues.asterisk.org/view.php?id=12713#c114447
----------------------------------------------------------------------
Most or all exits from the transfer logic seem to set needdestroy on the
SIP private structure, but a failed refer should not result in the call
going down and certainly is no guarantee that the calling party will clear
the call, so could end up with a stuck call on the calling party. Even the
dead leg of a successful transfer needs taking down cleanly, with a BYE,
which would typically come from the transferee.
Do I need to raise this as a new issue?
Issue History
Date Modified Username Field Change
======================================================================
2009-12-01 09:41 davidw Note Added: 0114447
======================================================================
More information about the asterisk-bugs
mailing list