[asterisk-bugs] [Asterisk 0018837]: [patch] Deadlock with attended transfer of SIP call
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Feb 21 02:33:31 CST 2011
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=18837
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Reported By: alecdavis
Assigned To:
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Project: Asterisk
Issue ID: 18837
Category: Core/RTP
Reproducibility: always
Severity: minor
Priority: normal
Status: ready for testing
Asterisk Version: 1.8.2.3
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2011-02-17 19:25 CST
Last Modified: 2011-02-21 02:33 CST
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Summary: [patch] Deadlock with attended transfer of SIP call
Description:
3 SIP phones.
A calls B, and B answers on line 1.
B puts A on hold by selecting line2.
B calls C, and C answers.
B initiates transfer of line1 to line2, phone uses REFER.
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Relationships ID Summary
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has duplicate 0018468 SIP crash on transfer
has duplicate 0018734 Combination dtmfmode=info, directmedia=...
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(0132217) Irontec (reporter) - 2011-02-21 02:33
https://issues.asterisk.org/view.php?id=18837#c132217
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Hi,
Yes I know RPID is different in two cases, but is Asterisk who sends two
Invites (with different RPID) and seems that some phones can't handle those
two consecutive Invites.
We've aplied the patch and this issue continues but it seems that the main
problem of deadlock is solved. (with directmedia=update and sipInfo)
Now I'm doing more probes...
Good job!
Issue History
Date Modified Username Field Change
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2011-02-21 02:33 Irontec Note Added: 0132217
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