[asterisk-bugs] [Asterisk 0014703]: sip call setup bug
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Apr 2 09:29:52 CDT 2009
The following issue has been CLOSED
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http://bugs.digium.com/view.php?id=14703
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Reported By: genie
Assigned To: dvossel
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Project: Asterisk
Issue ID: 14703
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: closed
Asterisk Version: 1.6.0.6
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
Resolution: open
Fixed in Version:
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Date Submitted: 2009-03-19 08:04 CDT
Last Modified: 2009-04-02 09:29 CDT
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Summary: sip call setup bug
Description:
I'm testing SIP protocol on Asterisk. While performing tests on performance
with packet loss I came across a bug of Asterisk while call initiation. In
my Asterisk SIP configuration I have 'canreinvite=no' set. That is why
normal call setup looks like that:
However, if the first ACK packet of the flow is lost and the ClientA who
does not know it sends Invite_with_authorisation packet the call will never
be set up, ending up with a infinte loop where client sends ACK messages
and asterisk '491 request pending' message. The wireshark files captured on
both ClientA and Asterisk can be found on bug.
regards,
genie
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(0102618) dvossel (administrator) - 2009-04-02 09:29
http://bugs.digium.com/view.php?id=14703#c102618
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committed a patch April 1st that should have taken care of this. Updating
to the latest branch code should fix it. If not, feel free to re-open this
issue and I'll take a closer look at it.
Issue History
Date Modified Username Field Change
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2009-04-02 09:29 dvossel Note Added: 0102618
2009-04-02 09:29 dvossel Status assigned => closed
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