[asterisk-bugs] [Asterisk 0018860]: [patch] Odd Behavior when dialed sip channel doesn't exist
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Apr 6 03:18:43 CDT 2011
The following issue has been UPDATED.
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https://issues.asterisk.org/view.php?id=18860
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Reported By: cmorford
Assigned To:
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Project: Asterisk
Issue ID: 18860
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: acknowledged
Asterisk Version: 1.8.2.3
JIRA:
Regression: No
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: 2011-02-21 10:56 CST
Last Modified: 2011-04-06 03:18 CDT
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Summary: [patch] Odd Behavior when dialed sip channel doesn't
exist
Description:
I found this previous issue that seems to be similar to mine:
https://issues.asterisk.org/view.php?id=18708
but the reporter was referred to forum. I believe the issue may be more
than a support issue.
On asterisk 1.6.1.13, the Dial command simply fails on a non-configured
extension and falls through the dialplan, terminating the call.
Using ChanIsAvail on the 1.6.1.13 system, I get an ${AVAILSTATUS} of 20,
which is expected.
On 1.8.2.3 the call stays inside the Dial command with the output below
and does not fail as expected. The ${AVAILSTATUS} is 0 which doesn't
appear to be appropriate.
If I stay on the phone, the __sip_xmit messages will repeat until I hang
up.
It interesting to note that the bogus extension I dialed was 166 which
shows up in the output below in the IP address.
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Relationships ID Summary
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related to 0018018 [patch] Hints for non-existent devices ...
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Issue History
Date Modified Username Field Change
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2011-04-06 03:18 schmidts Summary Odd Behavior when
dialed sip channel doesn't exist => [patch] Odd Behavior when dialed sip channel
doesn't exist
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