[asterisk-bugs] [Asterisk 0016238]: sip calls drop because of BYE's
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Jan 26 08:03:45 CST 2010
The following issue has been UPDATED.
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https://issues.asterisk.org/view.php?id=16238
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Reported By: seandarcy
Assigned To: file
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Project: Asterisk
Issue ID: 16238
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: closed
Asterisk Version: SVN
JIRA:
Regression: Yes
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
Resolution: fixed
Fixed in Version:
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Date Submitted: 2009-11-12 16:30 CST
Last Modified: 2010-01-26 08:03 CST
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Summary: sip calls drop because of BYE's
Description:
On 1.6.0.16, 1.6.0.18-rc1 and 2, SIP calls die because * sends a BYE. We
have 12 internal SIP phones. The server connects them to sip, iax and dahdi
over libpri. iax and dahdi work fine. The internal SIP phones work fine.
But outgoing SIP calls are terminated with a BYE. We've tried sip over both
Teliax and Junction. Same result. Teliax support told us they could see the
BYE.
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Relationships ID Summary
----------------------------------------------------------------------
related to 0016185 Crash of Outgoing Call
has duplicate 0016277 Outoing calls disconnected immediately ...
has duplicate 0016345 Call terminates 5 seconds after establi...
has duplicate 0016336 reinvites fail when sdp-session does no...
related to 0016676 1.6.2.1: sip BYE issued 90 seconds into...
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Issue History
Date Modified Username Field Change
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2010-01-26 08:03 lmadsen Target Version Potential Blocker =>
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