[asterisk-bugs] [Asterisk 0018837]: [patch] Deadlock with attended transfer of SIP call
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Feb 21 03:31:07 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 03:31 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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(0132220) alecdavis (manager) - 2011-02-21 03:31
https://issues.asterisk.org/view.php?id=18837#c132220
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Irontec: Now have a lookat https://issues.asterisk.org/view.php?id=18781.
If you have DAHDI/IAX trunks then transfer locally between SIP
extenstions, SEGFAULT
The patch there won't apply, but the lines like below can be manually
added.
- if (glue0->update_peer(c0, NULL, NULL, NULL, 0, 0)) {
+ if (glue0 != ast_rtp_instance_get_glue(c0->tech->type)) {
+ ast_log(LOG_WARNING, "Channel c0->'%s' technology changed, in bridge
with c1->'%s'\n", c0->name, c1->name);
+ } else if (glue0->update_peer(c0, NULL, NULL, NULL, 0, 0)) {
You get the idea.
Issue History
Date Modified Username Field Change
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2011-02-21 03:31 alecdavis Note Added: 0132220
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