[asterisk-bugs] [Asterisk 0014250]: Incoming calls matched to the wrong peer
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Jan 15 12:20:50 CST 2009
The following issue has been REOPENED.
======================================================================
http://bugs.digium.com/view.php?id=14250
======================================================================
Reported By: Nick_Lewis
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 14250
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: new
Asterisk Version: 1.6.0
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2009-01-15 05:58 CST
Last Modified: 2009-01-15 12:20 CST
======================================================================
Summary: Incoming calls matched to the wrong peer
Description:
If there are multiple sip trunks with the same ITSP then an incoming call
is arbitarily matched to the last peer with the same host IP address. This
is not a serious problem because the DID is still correct but it does have
many insidious effects due to the incorrect channel name
======================================================================
----------------------------------------------------------------------
(0097939) Nick_Lewis (reporter) - 2009-01-15 12:20
http://bugs.digium.com/view.php?id=14250#c97939
----------------------------------------------------------------------
The dial plan can compensate for a lot but as far as I am aware it cannot
compensate for the channel name. The channel name is assigned by
ast_channel_alloc in sip_new. The channel name is set to
SIP/<peerusername>-<instance>. If the wrong peer is matched it gets the
wrong channel name. The effects of this are subtle but things that rely on
channel state manager events such as the call state on the flash operator
panel are affected.
I am going to upload a patch that resolves this problem in many cases. It
will certainly not be to everyone's liking. It uses the approach of
ordinary UACs, such as sip phones, of matching the exten in the request URI
with the peername. It only does this after IP based authentication. If
there is no peername match then it drops back to the initial IP based match
so it is fully backward compatible.
Issue History
Date Modified Username Field Change
======================================================================
2009-01-15 12:20 Nick_Lewis Note Added: 0097939
2009-01-15 12:20 Nick_Lewis Status closed => new
2009-01-15 12:20 Nick_Lewis Resolution won't fix => reopened
======================================================================
More information about the asterisk-bugs
mailing list