[asterisk-bugs] [Asterisk 0018081]: SIP 200 OK withou ACK
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Oct 6 01:27:23 CDT 2010
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=18081
======================================================================
Reported By: marhbere
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 18081
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: SVN
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0
SVN Revision (number only!): 262803
Request Review:
======================================================================
Date Submitted: 2010-09-30 16:35 CDT
Last Modified: 2010-10-06 01:27 CDT
======================================================================
Summary: SIP 200 OK withou ACK
Description:
We have the issue when happening a problem into network.
When Receive INVITE and we respond 200 OK, and this last not arrived to
the originator, then the originator resend INVITE many times. But the
*Asterisk "Answers" all this call such as if it were differents calls.
I guess this is heavy problem when will try conciliation with CDR calls.
Please can tell me if we have any mistake
Tanks
======================================================================
----------------------------------------------------------------------
(0127732) schmidts (developer) - 2010-10-06 01:27
https://issues.asterisk.org/view.php?id=18081#c127732
----------------------------------------------------------------------
@Leif i understand what you mean, and asterisk acts as it should by
retransmit the 200 ok until T1*64 is reached and then drop the call but it
should handle the incoming invites (which are retransmits) not as a new
call.
i have looked at the code and the Ignoring this Invite message doesnt
cause anything to the dialog of this message.
@marhbere you could try to set "pedantic=yes" in the sip config. With this
enabled asterisk use a stricter method to find the old dialog and so this
may not happens.
Issue History
Date Modified Username Field Change
======================================================================
2010-10-06 01:27 schmidts Note Added: 0127732
======================================================================
More information about the asterisk-bugs
mailing list