[asterisk-bugs] [Asterisk 0016245]: [patch] [regression] The status of External SIP peer used as Queue member is not updating correctly
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Sep 14 07:48:02 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=16245
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Reported By: licedey
Assigned To:
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Project: Asterisk
Issue ID: 16245
Category: Applications/app_queue
Reproducibility: always
Severity: major
Priority: normal
Status: ready for testing
Asterisk Version: SVN
JIRA: SWP-597
Regression: Yes
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-11-13 23:49 CST
Last Modified: 2010-09-14 07:48 CDT
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Summary: [patch] [regression] The status of External SIP peer
used as Queue member is not updating correctly
Description:
The External SIP peer used in Queue doesn't update it's status when it
becomes available. The subscription hints are updated correctly but Queue
member status remaining in Unavailable state. When all queue members are
external peers, it ends up as all members are busy. It's possible to dial
directly external sip peer but not from the queue. The only way to correct
the status is to launch reload command from CLI.
This issue was discovers in April but still wasn't correct even with the
latest releases, here is the link to the discussion thread:
This issue is also discussed on FreePBX forum:
http://www.freepbx.org/forum/freepbx/installation/no-ringing-to-external-caller-with-queue
Here are the steps to reproduce the issue:
1. Configure external sip peers (qualify=yes) and add them to the Queue
2. Check the status the of sip peers and test the Queue
3. Remove internet connection from one of the external SIP. So the state
of sip peer/queue member is stated as Unavailable
4. Connect internet back
5. Now the subscription and peer state of the external sip peer should
become Idle/available again, but queue member state is still unavailable
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Relationships ID Summary
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related to 0012561 after agentdump, abandon message no lon...
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(0126923) gb_delti (reporter) - 2010-09-14 07:48
https://issues.asterisk.org/view.php?id=16245#c126923
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We are queuing calls to external providers that have only one number per
hotline and not one number per agent. So we definitely need the
member=SIP/1234 at myprovider notation.
Issue History
Date Modified Username Field Change
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2010-09-14 07:48 gb_delti Note Added: 0126923
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