[asterisk-bugs] [Asterisk 0007784]: Attended SIP Transfer Call Teardown Issue
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri May 27 10:45:10 CDT 2011
The following issue has been set as RELATED TO issue 0019372.
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https://issues.asterisk.org/view.php?id=7784
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Reported By: jcmoore
Assigned To:
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Project: Asterisk
Issue ID: 7784
Category: Channels/chan_sip/Transfers
Reproducibility: always
Severity: minor
Priority: normal
Status: closed
Asterisk Version: SVN
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 43322
Request Review:
Resolution: unable to reproduce
Fixed in Version:
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Date Submitted: 2006-08-22 20:45 CDT
Last Modified: 2011-05-27 10:45 CDT
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Summary: Attended SIP Transfer Call Teardown Issue
Description:
When attempting an attended SIP transfer which spans multilple Asterisk
servers, ie. INVITE w/ Replaces, the call on one of the phones does not
receive a BYE.
For example, in a call scenario from A->B->C, where A calls B, B then
attended transfers the call to C, the call on B is present even after the
call between A and C has been bridged. The audio is correctly bridged
between A and C, there is no audio on B.
The issue is that instead of deferring the BYE, as is currently being done
in handle_invite_replaces(), the BYE needs to proceed as normal so the call
on B will hangup while preserving the sip_pvt structure so that the INVITE
w/ Replaces can have a basis from which to create the new outbound INVITE.
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Relationships ID Summary
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duplicate of 0015833 Transfering phone left connected
related to 0019372 Attended transfer - transfering phone l...
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Issue History
Date Modified Username Field Change
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2011-05-27 10:45 lmadsen Relationship added related to 0019372
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