[asterisk-bugs] [Asterisk 0013438]: Call disconnection with branch in via header changing and scheduled destruction of sip call
Asterisk Bug Tracker
noreply at bugs.digium.com
Sun Sep 7 09:42:52 CDT 2008
The following issue has been SUBMITTED.
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http://bugs.digium.com/view.php?id=13438
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Reported By: agupta
Assigned To:
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Project: Asterisk
Issue ID: 13438
Category: Channels/chan_sip/General
Reproducibility: random
Severity: minor
Priority: normal
Status: new
Asterisk Version: 1.4.21.2
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 2008-09-07 09:42 CDT
Last Modified: 2008-09-07 09:42 CDT
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Summary: Call disconnection with branch in via header
changing and scheduled destruction of sip call
Description:
Randomly the call gets disconnected . The calls are coming on FXO card and
are routed to the IP Phone . The scenario is as --
1. INVITE is originated from asterisk to the IP Phone .
2. IP Phone sends , Trying , Ringing and 200OK
3. Asterisk responds with ACK and the branch in via header is different
from the INVITE/200OK
4. After 32 seconds asterisk puts the call for scheduled destruction and
the call is disconnected .
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Issue History
Date Modified Username Field Change
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2008-09-07 09:42 agupta Asterisk Version => 1.4.21.2
2008-09-07 09:42 agupta SVN Branch (only for SVN checkouts, not tarball
releases) => N/A
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